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
Requirements
- Asterisk 18 or later with PJSIP
- Shell access to Asterisk server
- Network reachability to sip.cliqtel.com
- At least one Cliqtel number
Trunk Configuration
When you connect, a SIP trunk is created with these optimized defaults:
Post-Setup Checklist
- pjsip show registrations shows Registered
- Inbound call reaches [from-cliqtel] context
- Outbound call routes via Cliqtel trunk
- No one-way audio (NAT traversal working)