asterisk srtp grandstream config

elected

Joined: Fri 17 of Dec, 2010

Re: asterisk srtp grandstream config


add to config peer

encryption=yes



700
type=friend
username=700
context=main
host=dynamic
secret=700
canreinvite=no
nat=yes
encryption=yes

701
type=friend
username=701
context=main
host=dynamic
secret=701
canreinvite=no
nat=yes
encryption=yes

Posted:Fri 17 of Dec, 2010 (11:05 UTC)
golgeyolcu

Joined: Tue 05 of Aug, 2008

asterisk srtp grandstream config


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.

Posted:Tue 05 of Aug, 2008 (12:26 UTC)
 
     

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