Asterisk-pjsip.conf: Difference between revisions
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md5_cred=51e63a3da6425a39aecc045ec45f1ae8 | md5_cred=51e63a3da6425a39aecc045ec45f1ae8 | ||
username=6001 | username=6001 | ||
</pre> | |||
=== AOR(Address of Record) === | |||
==== Example ==== | |||
Create automatic contact objects. | |||
<pre> | |||
[6001] | |||
type=aor | |||
max_contacts=1 | |||
</pre> | |||
Create manual contact objects. | |||
<pre> | |||
[6001] | |||
type=aor | |||
contact=sip:6001@192.0.2.1:5060 | |||
</pre> | |||
It's useful to note that you could define only the domain and omit the user portion of the SIP URI ifyou 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)" | |||
<pre> | |||
[mytrunk] | |||
type=aor | |||
contact=sip:203.0.113.1:5060 | |||
</pre> | </pre> | ||
Revision as of 14:06, 25 October 2017
Overview
Asterisk 의 pjsip 모듈 설정파일 pjsip.conf 내용 정리.
Basic
; Overview of Configuration Section Types Used in the Examples ; ; * Transport "transport" ; * Configures res_pjsip transport layer interaction. ; * Endpoint "endpoint" ; * Configures core SIP functionality related to SIP endpoints. ; * Authentication "auth" ; * Stores inbound or outbound authentication credentials for use by trunks, ; endpoints, registrations. ; * Address of Record "aor" ; * Stores contact information for use by endpoints. ; * Endpoint Identification "identify" ; * Maps a host directly to an endpoint ; * Access Control List "acl" ; * Defines a permission list or references one stored in acl.conf ; * Registration "registration" ; * Contains information about an outbound SIP registration ; * Phone Provisioning "phoneprov" ; * Contains information needed by res_phoneprov for autoprovisioning
ENDPOINT
TRANSPORT
TCP, UDP 혹은 WebSocket 과 같은 프로토콜이나 TLS/SSL 과 같은 암호화를 설정한다.
Example
A basic UDP transport bound to all interface.
[simpletrans] type=transport protocol=udp bind=0.0.0.0
TLS support, with many possible options and parameters.
[simpletrans] 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
접속 인증 설정.
Example
An example with username and password authentication.
[auth6001] type=auth auth_type=userpass password=6001 username=6001
MD5 authentication
[auth6001] type=auth auth_type=md5 md5_cred=51e63a3da6425a39aecc045ec45f1ae8 username=6001
AOR(Address of Record)
Example
Create automatic contact objects.
[6001] type=aor max_contacts=1
Create manual contact objects.
[6001] type=aor contact=sip:6001@192.0.2.1:5060
It's useful to note that you could define only the domain and omit the user portion of the SIP URI ifyou 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