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Odoo Phone settings

⚠️ WebRTC phone is supported in Google Chrome.

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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.

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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.

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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:
#exec /etc/asterisk/get_odoo_conf.sh
#tryinclude sip_odoo_auto_users.conf
  1. Script /etc/asterisk/get_odoo_conf.sh:
#!/bin/bash
curl -H "x-security-token: <server-security-token>" \
  http://localhost:8069/asterisk_plus/sip_peers > /tmp/sip_users.conf 2>/dev/null
if [ "$?" = "0" ]; then mv /tmp/sip_users.conf /etc/asterisk/sip_odoo_auto_users.conf ; fi
sleep 1
Reload Asterisk includes after updating.