Asterisk-pjsip.conf
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
type=transport
transport type. TCP, UDP 혹은 WebSocket 과 같은 프로토콜이나 TLS/SSL 과 같은 암호화를 설정한다.
Example
Basic UDP transport
[transport-udp] type=transport protocol=udp ;udp,tcp,tls,ws,wss bind=0.0.0.0
UDP transport behind NAT
[transport-udp-nat] type=transport protocol=udp bind=0.0.0.0 local_net=192.0.2.0/24 external_media_address=203.0.113.1 external_signaling_address=203.0.113.1
Basic IPv6 UDP tranport
[transport-udp-ipv6] type=transport protocol=udp bind=::
Exmple IPv4 TLS transport
[transport-tls] type=transport protocol=tls bind=0.0.0.0 cert_file=/path/mycert.crt priv_key_file=/path/mykey.key cipher=ADH-AES256-SHA,ADH-AES128-SHA method=tlsv1
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