[Asterisk-Users] Is SIP BROKEN?

[Paul Mahler]

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?