SIP Configuration
This guide explains how to configure SIP phones and softphones to work with OPBX User extensions.
What is SIP?
Session Initiation Protocol (SIP) is the standard protocol for voice over IP (VoIP) communications. It allows phones to register with OPBX and make/receive calls over the internet.
Only User Extensions have SIP credentials. Other extension types (Conference, Ring Group, etc.) do not use SIP.
SIP Credentials
Each User Extension has unique SIP credentials:
| Field | Value | Example |
|---|---|---|
| Username | Extension number | 101 |
| Password | 32-character hex string | a1b2c3d4e5f6... |
| Server/Domain | Your Cloudonix domain | yourdomain.cloudonix.net |
| Transport | UDP or TCP | UDP (recommended) |
Finding Your SIP Credentials
- Navigate to Extensions
- Find your User extension
- Click View Details or SIP Settings
- Note the:
- Extension number (username)
- SIP password
- Server address
SIP Password Security:
- Passwords are 32-character secure strings
- Passwords are hidden from normal API responses
- Viewing passwords is logged for audit
- Regenerate passwords if compromised
Phone Configuration
IP Desk Phones
Most IP phones support SIP configuration through their web interface or menu system.
Generic Configuration Steps
-
Access phone settings:
- Web interface: Enter phone's IP address in browser
- Menu: Press Settings/Menu button on phone
-
Navigate to SIP/Account settings
-
Enter credentials:
- Account Name/Label: Any descriptive name (e.g., "John's Desk")
- SIP User ID: Extension number (e.g.,
101) - SIP Password: The 32-character password
- SIP Server/Domain: Your Cloudonix domain
- Display Name: Name shown to callers (e.g., "John Smith")
-
Save and restart the phone
-
Verify registration:
- Look for "Registered" status
- Test by making a call
Popular Phone Models
Yealink
Settings → SIP Account → Account 1:
- Line Active: Enabled
- Label: John Smith
- Display Name: John Smith
- Register Name: 101
- User Name: 101
- Password: [32-char password]
- SIP Server: yourdomain.cloudonix.net
- Transport: UDP
Cisco/Linksys SPA
Voice → Ext 1:
- Line Enable: yes
- SIP Port: 5060
- Proxy: yourdomain.cloudonix.net
- Display Name: John Smith
- User ID: 101
- Password: [32-char password]
- Preferred Codec: G711u
Grandstream
Accounts → Account 1 → General Settings:
- Account Name: John Smith
- SIP Server: yourdomain.cloudonix.net
- SIP User ID: 101
- SIP Password: [32-char password]
- Name: John Smith
Softphones
Softphones are applications that run on computers or mobile devices.
Desktop Softphones
MicroSIP (Windows):
- Account Name: John Smith
- SIP Server: yourdomain.cloudonix.net
- Username: 101
- Domain: yourdomain.cloudonix.net
- Password: [32-char password]
- Transport: UDP
Zoiper (Windows/Mac/Linux):
- Username: 101
- Password: [32-char password]
- Domain: yourdomain.cloudonix.net
- Transport: UDP
Telephone (Mac):
- Full Name: John Smith
- Domain: yourdomain.cloudonix.net
- Username: 101
- Password: [32-char password]
Mobile Softphones
Zoiper (iOS/Android):
- Account Name: John Smith
- Host: yourdomain.cloudonix.net
- Username: 101
- Password: [32-char password]
- Transport: UDP
Groundwire (iOS):
- Display Name: John Smith
- Username: 101
- Password: [32-char password]
- Domain: yourdomain.cloudonix.net
Advanced Configuration
Codec Selection
Recommended codec order:
- G.711 (μ-law) - Best quality, higher bandwidth
- G.711 (a-law) - Best quality, higher bandwidth
- G.729 - Good quality, lower bandwidth
NAT/ Firewall Considerations
If phones are behind NAT (most home/office networks):
- STUN Server: May be needed for NAT traversal
- Keep-Alive: Enable to maintain registration
- Local SIP Port: Can use 5060 or random high port
QoS (Quality of Service)
For best call quality:
- Enable QoS on your router if available
- Prioritize RTP (voice) traffic
- Set DSCP: EF (Expedited Forwarding) for voice
Troubleshooting
Phone Won't Register
Problem: Phone shows "Not Registered" or "Registration Failed"
Solutions:
-
Verify credentials:
- Double-check extension number
- Verify password (copy/paste to avoid typos)
- Confirm server/domain name
-
Check network:
- Ensure phone has internet access
- Verify firewall isn't blocking SIP (port 5060)
- Try different transport (UDP vs TCP)
-
Check extension status:
- Verify extension is "Active" in OPBX
- Ensure extension is assigned to a user
- Check if extension is already registered elsewhere
-
Review logs:
- Check phone's registration logs
- Look for authentication errors
- Verify DNS resolution of server name
Calls Drop Frequently
Problem: Calls disconnect unexpectedly
Solutions:
- Check network stability
- Adjust codec (try G.711 instead of G.729)
- Enable/disable STUN based on NAT setup
- Check keep-alive settings
- Review QoS configuration
One-Way Audio
Problem: Can hear but not be heard (or vice versa)
Solutions:
-
NAT/Firewall issue:
- Enable STUN on phone
- Check router SIP ALG settings
- Try different local SIP ports
-
Codec mismatch:
- Ensure both sides support common codec
- Set explicit codec order
-
RTP port issues:
- RTP ports typically 10000-20000
- Ensure firewall allows RTP traffic
Poor Call Quality
Problem: Choppy audio, delays, or echo
Solutions:
-
Bandwidth:
- G.711 needs ~100 kbps per call
- Check available bandwidth
- Prioritize voice traffic with QoS
-
Jitter/Packet Loss:
- Check network quality
- Use jitter buffer on phone
- Switch to wired connection if on WiFi
-
Echo:
- Enable echo cancellation on phone
- Reduce microphone sensitivity
- Use headset instead of speakerphone
Security Best Practices
Password Management
- Never share SIP passwords - Each user should have their own
- Regenerate if compromised - Change passwords if you suspect a leak
- Use strong passwords - System generates secure passwords automatically
- Regular rotation - Change passwords periodically
Network Security
-
Use strong WiFi passwords - Prevent unauthorized network access
-
Enable phone security features:
- Admin password on phone web interface
- Disable unused services
- Keep phone firmware updated
-
VPN for remote users - Consider VPN for users working remotely
Monitoring
- Review call logs regularly
- Check for unusual activity:
- Calls at odd hours
- Unexpected international calls
- High volume of short calls
- Audit SIP password access - Review who has viewed passwords
Regenerating SIP Passwords
If a password is compromised or needs changing:
- Navigate to Extensions
- Find the User extension
- Click Regenerate Password or Reset SIP Credentials
- Update all devices with the new password
- Verify registration on each device
Regenerating a password will disconnect all devices using the old password. Update devices immediately.
Multiple Devices
A single User extension can typically register multiple devices (desk phone + softphone).
Benefits:
- Ring multiple devices simultaneously
- Flexibility to use different devices
Considerations:
- All devices ring on incoming calls
- Any device can answer
- All devices share the same voicemail
Next Steps
- Extension Types - Learn about different extension types
- Managing Extensions - Edit and maintain extensions
- Troubleshooting - General troubleshooting guide
Related Documentation:
- Extension Assignment - Link users to extensions
- Cloudonix Integration - Understanding the backend