[Asterisk-Users] Is SIP BROKEN?

Eric Wieling eric at fnords.org
Sat Apr 24 12:34:12 MST 2004


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? 
> 
> 
> 
> 
> 
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-- 
          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."




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