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
