I changed the numbers but it didn't work. On Thu, Aug 7, 2008 at 5:45 AM, golge yolcu <[EMAIL PROTECTED]>wrote:
> 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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