[Asterisk-Users] Is SIP BROKEN?

[Karl Brose]

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 at signate.com>
To: <asterisk-users at 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?
>
>
>
>
>
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