in sip.conf [general] port = 5060 ; The TCP/IP port for SIP communiations bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses on server. context=other ; Default for incoming calls disallow=all allow=ulaw allow=gsm in extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here [globals] [inside] exten => 77,1,voicemailmain [other] exten => 88,1,Playback(demo-congrats) Next, I have an x-lite phone set up as Display name: 40 Username: 40 Authorization user: 40 Domain/Realm: 69.240.152.95 SIP Proxy: 69.240.152.95 I get a message from SIP debug that says 40 from the x-lite is failing to register. This should be the case since I don't have any sip entry for 40. Here's the weird part. If I dial 77 from the x-lite phone I get sent to voice mail. If I dial 88 from the x-lite phone I get the demo-congrats message. Why am I getting anything? Why aren't these calls failing?
I'm sure you know that if there is no matching [sipentry] for an incoming call it will be allowed thru and will take it's settings from [general] in sip.conf. Try setting something like context=INVALID in [general] and then set a context= line for each [sipentry]. That way if a connection doesn't match a [sipentry] it will fail, rather than be accepted with all the default settings. Also see http://www.fnords.org/~eric/asterisk/sip.conf.shtml On Sat, 2004-04-24 at 14:26, Paul Mahler wrote:> in sip.conf > [general] > port = 5060 ; The TCP/IP port for SIP communiations > bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses > on server. > context=other ; Default for incoming calls > disallow=all > allow=ulaw > allow=gsm > > in extensions.conf > [general] > static=yes ; These two lines prevent the command-line interface > writeprotect=yes ; from overwriting the config file. Leave them here > [globals] > > [inside] > exten => 77,1,voicemailmain > > [other] > exten => 88,1,Playback(demo-congrats) > > > Next, I have an x-lite phone set up as > Display name: 40 > Username: 40 > Authorization user: 40 > Domain/Realm: 69.240.152.95 > SIP Proxy: 69.240.152.95 > > > I get a message from SIP debug that says 40 from the x-lite is failing to > register. This should be the case since I don't have any sip entry for 40. > > Here's the weird part. If I dial 77 from the x-lite phone I get sent to > voice mail. If I dial 88 from the x-lite phone I get the demo-congrats > message. Why am I getting anything? Why aren't these calls failing? > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."
Asterisk will accept unauthenticated calls, defaulting to the context specified in the general section. Therefore only the call to extension 88 should work. If both, 77 and 88, are working for you then, yes, something is broken. ----- Original Message ----- From: "Paul Mahler" <pmahler@signate.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, April 24, 2004 15:26 Subject: [Asterisk-Users] Is SIP BROKEN?> in sip.conf > [general] > port = 5060 ; The TCP/IP port for SIP communiations > bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses > on server. > context=other ; Default for incoming calls > disallow=all > allow=ulaw > allow=gsm > > in extensions.conf > [general] > static=yes ; These two lines prevent the command-line interface > writeprotect=yes ; from overwriting the config file. Leave them here > [globals] > > [inside] > exten => 77,1,voicemailmain > > [other] > exten => 88,1,Playback(demo-congrats) > > > Next, I have an x-lite phone set up as > Display name: 40 > Username: 40 > Authorization user: 40 > Domain/Realm: 69.240.152.95 > SIP Proxy: 69.240.152.95 > > > I get a message from SIP debug that says 40 from the x-lite is failing to > register. This should be the case since I don't have any sip entry for 40. > > Here's the weird part. If I dial 77 from the x-lite phone I get sent to > voice mail. If I dial 88 from the x-lite phone I get the demo-congrats > message. Why am I getting anything? Why aren't these calls failing? > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users