Asterisk pjsip configuration: Difference between revisions

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Reloading Config: Configuration for transport type sections can't be reloading 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.
Reloading Config: Configuration for transport type sections can't be reloading 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.
A basic UDP transport bound to all interfaces
<pre>
<pre>
; A basic UDP transport bound to all interfaces
[simpletrans]
[simpletrans]
type=transport
type=transport
protocol=udp
protocol=udp
bind=0.0.0.0
bind=0.0.0.0
</pre>


; Or a TLS transport, with many possible options and parameters
Or a TLS transport, with many possible options and parameters
[simpletrans]
<pre>
[simpletranst]
type=transport
type=transport
protocol=tls
protocol=tls

Revision as of 23:41, 1 December 2018

Overview

Asterisk pjsip configuration.

Configuration format

[  SectionName ] 
ConfigOption = Value 
ConfigOption = Value

Section names

대부분의 경우, 섹션 이름은 아무렇게나 지정할 수 있다. 예를 들어 transport 이름을 [transport-udp-nat] 와 같이 기억하기 쉽게 지정할 수도 있다.

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 "from" header for inbound SIP request. 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 "From" header.

Section types

Below is 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.

There are dozens of config options for some of the sections, but the examples below are very minimal for the sake of simplicity.

ENDPOINT

provided by module: res_pjsip

Endpoint configuration provides numerous options relating to core SIP functionality and ties to other sections such as auth, aor and transport. You can't contact an endpoint without associating one or more AoR sections. An endpoint is essentially a profile for the configuration of a SIP endpoint such as a phone or remote server.

[6001]
type=endpoint
context=default
disallow=all
allow=ulaw
transport=simpletrans
auth=auth6001
aors=6001

If you want to define the Caller id this endpoint should use, then add something like the following.

trust_id_outbound=yes
callerid=Spaceman Spiff <6001>

TRANSPORT

provided by module: res_pjsip

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 reloading 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.

A basic UDP transport bound to all interfaces

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

Or a TLS transport, with many possible options and parameters

[simpletranst]
type=transport
protocol=tls
bind=0.0.0.0
;various TLS specific options below:
cert_file=
priv_key_file=
ca_list_file=
cipher=
method=