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 at 172.16.70.121
register => 102:102 at 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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Leandro Dardini
2012-Mar-26 07:44 UTC
[asterisk-users] Settings problems of Asterisk as client
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_ch1987 at 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 at 172.16.70.121 > register => 102:102 at 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 > *** > * > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120326/443ade74/attachment.htm>