Your problem originate from the use of insecure option. Using this option,
the peer is authenticated using the registration ip and not the user and
password.
Leandro
Il giorno 26/mar/2012 05:48, "YeungJoe" <ma_ch1...@hotmail.com> ha scritto:
> Hello All,
>
> I am Asterisk user, and right now I have some troubles about Asterisk As
> Client settings.
>
> Here are my envrionments:
>
> Asterisk-1.8.5.0
>
> -----------------------------------------------------------
> Server Settings(IP:172.16.70.121)
>
> ////////////extensions.conf////////////////
>
>
> [from-internal-200]
> exten => _X.,1,Dial(SIP/${EXTEN})
> exten => _X.,n,Hangup()
>
> ////////////end of extensions.conf/////////
>
>
> ////////////sip.conf///////////////////////
> [101]
> type=friend
> username=101
> secret=101
> host=dynamic
> allow=all
> context=from-internal-101
>
>
> [102]
> type=friend
> username=102
> secret=102
> host=dynamic
> allow=all
> context=from-internal-102
>
>
> [200]
> type=friend
> username=200
> secret=200
> host=dynamic
> allow=all
> context=from-internal-200
> ////////////////////////end of sip.conf///////////
>
> -----------------------------------------------------------
> Client Settings(IP:172.16.70.124:
>
> //////////////////////extensions.conf//////////
> [from-sip-101]
> exten => s,1,Noop(SIP-101)
>
> [from-sip-102]
> exten => s,1,Noop(SIP-102)
> ////////////////////end of extensions.conf/////
>
>
> /////////////////////sip.conf//////////////////
> [general]
> register => 101:101@172.16.70.121
> register => 102:102@172.16.70.121
>
> [101]
> type=peer
> username=101
> secret=101
> insecure=invite,port
> host=172.16.70.121
> context=from-sip-101
>
> [102]
> type=peer
> username=102
> secret=102
> insecure=invite,port
> host=172.16.70.121
> context=from-sip-102
> //////////////////end of sip.conf/////////////
> -----------------------------------------------------------
>
> Right now, I am able to register extensions 101 and 102 to
> server(172.16.70.121).
> and I can dial from SIP extension 200 to 101 or 102, if I dial 101, it
> will be
> routed to 101, and 101 is ringing. This is OK. but if I dial 102, it also
> be routed 101, I don't know why, because
> according to my SIP knowledges it should be routed to 102 as they are
> different contexts.
>
> BTW, Client peer is also based on Asterisk.
>
> I am a newbie of SIP, if you need more info I will provide.
> Please help! Thanks!
>
>
> Joe.Yeung
> ***
> *
>
>
>
> --
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