Skip to main content

Odoo Phone settings

⚠️ WebRTC phone is supported in Google Chrome.

System settings (PBX → Settings)

  • Set Asterisk SIP URI, WebSocket address, STUN server; adjust ports if you changed defaults (SIP 5060, HTTPS 8089).
  • Transfer contact search: all contacts, PBX users, or both.
  • Attended Transfer Sequence / Disconnect Call Sequence: must match your Asterisk features.conf.
  • Enable Generate SIP peers to manage SIP accounts in Odoo and push them to Asterisk.

User setup

  • Create PBX User mapping with a WebRTC channel; refresh UI to show the phone icon.
  • Set Auto Answer Header to Answer-Mode: Auto for better UX.

Auto-provision SIP users to Asterisk

  1. Enable Generate SIP peers in PBX → Settings → Server → SIP Users.
  2. In asterisk.conf set execincludes = yes.
  3. Include generated peers in pjsip_wizard.conf:
  1. Script /etc/asterisk/get_odoo_conf.sh:
Reload Asterisk includes after updating.