Asterisk pjsip configuration
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.
Example
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=
AUTH
provided by module: res_pjsip
Authentication sections hold the options and credentials related to inbound or outbound authentication. You'll associate other section such as endpoints or registrations to this one.
Multiple endpoints or registrations can use a single auth config if needed.
Example
An example with username and password authentication
[auth6001] type=auth auth_type=userpass password=6001 username=6001
And then an example with MD5 authentication.
[auth6001] type=auth auth_type=md5 md5_cred=51e63a3da6425a39aecc045ec45f1ae8 username=6001
AOR
provided by module: res_pjsip
A primary feature of AOR objects(Address of Record) is to tell Asterisk where an endpoint can be contacted. Without an associated AOR section, an endpoint can not be contacted. AOR objects also store associations to mailboxes for MWI requests and other data that might relate to the whole group of contacts such as expiration and qualify settings.
When Asterisk receives an inbound registration, it'll look to match against available AORs.
Registration: The name of the AOR section must match the user portion of the SIP URI in the "To:" header of the inbound SIP registration. That will usually be the "user name" set in you hard or soft phone configuration.
Example
First, we have a configuration where you are expecting the SIP User Agent (likely a phone) to register against the AOR. In this case, the contact objects will be created automatically. We limit the maximum contact creation to 1. We could do 10 if we wanted up to 10 SIP User Agents to be able to register against it.
[6001] type=aor max_contacts=1
Second, we have a configuration where you are not expecting the SIP User Agent to register against the AOR. In this case, you can assign contacts manually as follows. We don't have to worry about max_contacts since that option only affects the maximum allowed contacts to be created through external interaction, like registration.
[6001] type=aor contact=sip:6001@192.0.2.1:5060
Third, it's useful to note that you could define only the domain and omit the user portion of the SIP URI if you wanted. Then you could define the user portion dynamically in your dialplan when calling the Dial application. You'll likely do this when building an AOR/Endpoint combo to use for dialing out to an ITSP. For example: "Dial(PJSIP/${EXTEN}@mytrunk)"
[mytrunk] type=aor contact=sip:203.0.113.1:5060
REGISTRATION
provided by module: res_pjsip_outbound_registration
The registration section contains information about an outbound registration. You'll use this when setting up a registration to another system whether it's local or a trunk from your ITSP.
Example
This example shows you how you might configure registration and outbound authentication against another Asterisk system, where the other system is using the older cahn_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 call, 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.
[mytrunk] type=registration transport=simpletrans outbound_auth=mytrunk server_uri=sip:myaccountname@203.0.113.1:5060 client_uri=sip:myaccountname@192.0.2.1:5060 retry_interval=60
And an example that may work with SIP trunking provider.
[mytrunk] type=registration transport=simpletrans outbound_auth=mytrunk server_uri=sip:sip.example.com client_uri=sip:1234567890@sip.example.com retry_interval=60
What if you don't need to authenticate? You can simply omit the outbound_auth option.
IDENTIFY
provided by module: res_pjsip_endpoint_identifier_ip
Controls how the res_pjsip_endpoint_identifier_ip module determines what endpoint an incoming packet is from. If you don't have an identify section defined, or else you have res_pjsip_endpoint_identifier_ip loading after res_pjsip_endpoint_identifier_user, then res_pjsip_endpoint_identifier_user will identify inbound traffic by pulling the user from the "from:" SIP header in the packet. Basically the module load order, and your configuration will both determine whether you identify by IP or by user.
Example
Its use is quite straightforward. With this configuration if Asterisk sees inbound traffic from 203.0.113.1 then it will match that to Endpoint 6001.
[6001] type=identify endpoint=6001 match=203.0.113.1
CONTACT
provided by module: res_pjsip
The contact config object effectively acts as an