However, in some cases, (endpoint and aor types) the section name has a relationship to its function. In the case of endpoint and aor their names must match the user portion of the SIP URI in the "ToFrom" header for inbound SIP requests. The exception to that rule is if you have an identify section configured for that endpoint. In that case the inbound request would be matched by IP instead of against the user in the "ToFrom" header.
Below is a brief description of each section type and an example showing configuration of that section only. The module providing the configuration object related to the section is listed in parentheses next to each section name.
Configure how res_pjsip will operate at the transport layer. For example, it supports configuration options for protocols such as TCP, UDP or WebSockets and encryption methods like TLS/SSL.
You can setup multiple transport sections and other sections (such as endpoints) could each use the same transport, or a unique one. However, there are a couple caveats for creating multiple transports:
- They cannot share the same IP+port or IP+protocol combination. That is, each transport that binds to the same IP as another must use a different port or protocol.
- PJSIP does not allow multiple TCP or TLS transports of the same IP version (IPv4 or IPv6).
Reloading Config: Configuration for transport type sections can't be reloaded during run-time without a full module unload and load. You'll effectively need to restart Asterisk completely for your transport changes to take effect.
This example shows you how you might configure registration and outbound authentication against another Asterisk system, where the other system is using the older chan_sip peer setup.
This example is just the registration itself. You'll of course need the associated transport and auth sections. Plus, if you want to receive calls from the far end (who now knows where to send calls, thanks to your registration!) then you'll need endpoint, AOR and possibly identify sections setup to match inbound calls to a context in your dialplan.
And an example that may work with a SIP trunking provider
What if you don't need to authenticate? You can simply omit the outbound_auth option.
(provided by module: res_pjsip)
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