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wiki:voip:asterisk:sip_trunk

SIP Trunk

Server A

sip.conf
[serverA]
type = peer
host = 192.168.1.101
username = serverB
secret = apples
context = incoming
disallow = all
allow = alaw
extensions.conf
exten => _5XXX,1,Dial(SIP/${EXTEN}@serverA)

Server B

sip.conf
[serverB]
type = peer
host = 192.168.1.102
username = serverA
secret = apples
context = incoming
disallow = all
allow = alaw
extensions.conf
exten => _6XXX,1,Dial(SIP/${EXTEN}@serverB)

Trunk s registrací

sip.conf
[general]
...
register => username:password@your.provider.tld
...

[myprovider]
type = peer
host = your.provider.tld
username = username
secret = password
; Most providers won't authenticate when they send calls to you,
; so you need this line to just accept their calls.
insecure = invite
dtmfmode = rfc2833
disallow = all
allow = alaw
extensions.conf
exten => _XXXXXXXXX,1,Dial(SIP/${EXTEN}@myprovider)

Šifrování hovorů (SIP TLS, SRTP)

FIXME str.150

wiki/voip/asterisk/sip_trunk.txt · Last modified: 2014/12/26 18:31 (external edit)