I changed the numbers but it didn't work.
On Wed, Aug 6, 2008 at 8:40 AM, Peter P GMX <[EMAIL PROTECTED]> 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. > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-security mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-security > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-security mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-security >
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