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Chapter 3 — Setup

SIP & PBX Configuration

Snorlax, the configuration editor for the software PBX module, and the SIP trunk parameters required to connect the platform to external telephony infrastructure.

The Snorlax module

Snorlax provides a centralised interface for managing and deploying the configuration of the software-based PBX. Administrators edit PBX configuration files, manage SIP connectivity and control dialplan behaviour without direct access to the telephony server or command line.

After a secure connection to the PBX host is established, users can browse and modify core Asterisk configuration files such as extensions.conf , extensions.ael , pjsip.conf and sip.conf . The integrated editor allows real-time modification of routing logic, call handling rules and integration points with the VoiceBot platform; once validated, updated configurations are deployed directly from the interface.

Non-intrusive deployment

Configuration updates do not affect ongoing calls. New settings apply only to calls initiated after the changes have been saved and deployed.

Main functionality

  • SIP parameter configuration — SIP server, port, user credentials, password and protocol (UDP/TCP/TLS).
  • Direct editing of configuration files — view and edit the exchange's .conf files without SSH access.
  • Trunk management — configure SIP trunks for external providers, enterprise PBX systems or cloud telephony platforms, including trunk parameters, routing rules and connection endpoints for inbound and outbound calls.
  • Integration with the VoiceBot engine — the configured SIP parameters are used to establish both incoming and outgoing calls.
  • Apply changes (deploy) — save and immediately activate new settings without restarting the system.

Figure 4 — Snorlax interface

Figure 4 — Snorlax interface.

SIP trunk requirements

The Software PBX connects to external telephony infrastructure through a SIP trunk configured in the Asterisk PJSIP stack and managed through Snorlax. The configuration covers the SIP endpoint address, transport protocol, authentication credentials, registration behaviour, codec negotiation and inbound/outbound routing contexts. The PJSIP wizard profile defines trunk type, authentication method, remote host parameters and the dialplan context used for incoming and outgoing calls; additional parameters specify NAT traversal, media transport and allowed codecs.

Parameter Description
type Configuration object type — typically wizard for simplified PJSIP trunk configuration.
remote_hosts IP address or domain of the remote SIP provider or external PBX.
transport SIP signalling transport protocol: udp, tcp or tls.
username Username used for SIP authentication with the telephony provider.
password Authentication password associated with the SIP trunk account.
server_uri SIP URI of the provider used for registration and signalling.
client_uri SIP URI identifying the OpenBrain PBX during registration.
context Dialplan context where incoming calls from the SIP trunk are routed.
allow Permitted audio codecs for media negotiation, for example ulaw, alaw or opus.
outbound_proxy Optional proxy server used for routing SIP requests through the provider network.

Because this page controls the parameters that define communication between the voice platform and the telephony environment, it is intended for administrators and system engineers.