Features
- Direct PJSIP object configuration
- Full dialplan control (extensions.conf)
- Supports all Asterisk codecs including Opus
- CLI-based validation and SIP tracing
- Works with Asterisk 18+ and PJSIP
How to Connect
1
Create PJSIP objects
Add auth, AOR, and endpoint objects to your pjsip.conf (or pjsip_custom.conf) using the credentials below.
2
Add dialplan context
Create a [from-didfarm] context in extensions.conf to handle inbound calls from DIDfarm numbers.
3
Reload and validate
Run "pjsip reload" from the Asterisk CLI. Check registration with "pjsip show registrations".
4
Run test traffic
Call your DIDfarm number and verify it hits the correct dialplan context. Check SIP traces if needed.
Requirements
- Asterisk 18 or later with PJSIP
- Shell access to Asterisk server
- Network reachability to sip.didfarm.com
- At least one DIDfarm number
Trunk Configuration
When you connect, a SIP trunk is created with these optimized defaults:
transport
udp
codec preference
ulaw,alaw,g722,opus
nat mode
auto
auth type
credentials
max channels
4
Post-Setup Checklist
- pjsip show registrations shows Registered
- Inbound call reaches [from-didfarm] context
- Outbound call routes via DIDfarm trunk
- No one-way audio (NAT traversal working)