<div dir="ltr">I changed the numbers but it didn't work.<br><br><div class="gmail_quote">On Thu, Aug 7, 2008 at 5:45 AM, golge yolcu <span dir="ltr"><<a href="mailto:golgeyolcu.uye@googlemail.com">golgeyolcu.uye@googlemail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div dir="ltr">I changed the numbers but it didn't work.<div><div></div><div class="Wj3C7c">
<br><br><br><div class="gmail_quote">On Wed, Aug 6, 2008 at 8:40 AM, Peter P GMX <span dir="ltr"><<a href="mailto:Prometheus001@gmx.net" target="_blank">Prometheus001@gmx.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Maybe you should not use the same numbers (700, 701) in your dialplan as<br>
for your extensions.<br>
<br>
Best regards<br>
Peter<br>
<br>
golge yolcu schrieb:<br>
<div><div></div><div>><br>
> Asterisk SRTP config<br>
><br>
> i installed asterisk with srtp. i have configured sip.conf and<br>
> extensions.conf like<br>
><br>
> extensions.conf<br>
> main<br>
> exten => 600,1,Set(_SIPSRTP=optional)<br>
> exten => 600,n,Set(_SIPSRTP_CRYPTO=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<br>
> succesful and i am getting my echo but when i dial 700 it says call<br>
> failed reason code : 603<br>
><br>
> Is there anybody who can help me.<br>
</div></div>> ------------------------------------------------------------------------<br>
><br>
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</blockquote></div><br></div></div></div>
</blockquote></div><br></div>