[asterisk-security] asterisk srtp config problem

[golge yolcu]

I changed the numbers but it didn't work.


On Wed, Aug 6, 2008 at 8:40 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> 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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