[asterisk-security] asterisk srtp config problem
golge yolcu
golgeyolcu.uye at googlemail.com
Tue Aug 5 08:43:25 CDT 2008
Asterisk SRTP config
i installed asterisk with srtp. i have configured sip.conf and
extensions.conf like
extensions.conf
main
exten => 600,1,Set(_SIPSRTP=optional)
exten => 600,n,Set(_SIPSRTP_CRYPTO=enable)
exten => 600,n,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,hangup
exten => 610,1,Set(_SIPSRTP=require)
exten => 610,n,Set(_SIPSRTP_MIKEY=enable)
exten => 610,n,Playback(demo-echotest) ; Let them know what's going on
exten => 610,n,Echo ; Do the echo test
exten => 610,n,Playback(demo-echodone) ; Let them know it's over
exten => 610,n,hangup
exten => 700, 1, Set(_SIP_SRTP_SDES=1)
exten => 700, n, Set(_SIPSRTP=optional)
exten => 700, n, Set(_SIPSRTP_CRYPTO=enable)
exten => 700, n, Dial(SIP/700)
exten => 701, 1, Set(_SIP_SRTP_SDES=1)
exten => 701, n, Set(_SIPSRTP=optional)
exten => 701, n, Set(_SIPSRTP_CRYPTO=enable)
exten => 701, n, Dial(SIP/701)
sip.conf
700
type=friend
username=700
context=main
host=dynamic
secret=700
canreinvite=no
nat=yes
701
type=friend
username=701
context=main
host=dynamic
secret=701
canreinvite=no
nat=yes
and i used grandstream GXP2020 telephones. when i dial 600 it is succesful
and i am getting my echo but when i dial 700 it says call failed reason code
: 603
Is there anybody who can help me.
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