<div dir="ltr">Asterisk SRTP config
<br>
<br>i installed asterisk with srtp. i have configured sip.conf and extensions.conf like
<br>
<br>extensions.conf
<br>main
<br>exten => 600,1,Set(_SIPSRTP=optional)
<br>exten => 600,n,Set(_SIPSRTP_CRYPTO=<div dir="ltr">enable)
<br>exten => 600,n,Playback(demo-echotest) ; Let them know what's going on
<br>exten => 600,n,Echo ; Do the echo test
<br>exten => 600,n,Playback(demo-echodone) ; Let them know it's over
<br>exten => 600,n,hangup
<br>
<br>exten => 610,1,Set(_SIPSRTP=require)
<br>exten => 610,n,Set(_SIPSRTP_MIKEY=enable)
<br>exten => 610,n,Playback(demo-echotest) ; Let them know what's going on
<br>exten => 610,n,Echo ; Do the echo test
<br>exten => 610,n,Playback(demo-echodone) ; Let them know it's over
<br>exten => 610,n,hangup
<br>
<br>
<br>exten => 700, 1, Set(_SIP_SRTP_SDES=1)
<br>exten => 700, n, Set(_SIPSRTP=optional)
<br>exten => 700, n, Set(_SIPSRTP_CRYPTO=enable)
<br>exten => 700, n, Dial(SIP/700)
<br>
<br>exten => 701, 1, Set(_SIP_SRTP_SDES=1)
<br>exten => 701, n, Set(_SIPSRTP=optional)
<br>exten => 701, n, Set(_SIPSRTP_CRYPTO=enable)
<br>exten => 701, n, Dial(SIP/701)
<br>
<br>sip.conf
<br>
<br>700
<br>type=friend
<br>username=700
<br>context=main
<br>host=dynamic
<br>secret=700
<br>canreinvite=no
<br>nat=yes
<br>
<br>701
<br>type=friend
<br>username=701
<br>context=main
<br>host=dynamic
<br>secret=701
<br>canreinvite=no
<br>nat=yes
<br>
<br>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
<br>
<br>Is there anybody who can help me.</div></div>