[asterisk-security] asterisk srtp config problem

[Peter P GMX]

Maybe you should not use the same numbers (700, 701) in your dialplan as 
for your extensions.

Best regards
Peter

golge yolcu schrieb:
>
> 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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