[Asterisk-Users] Sip Registration Problem

[Karl Brose]

No and Yes, Olle. But mostly NO.
What Asterisk is doing actually depends on how it is configured.  If you 
are, by design, accepting calls for a particular [user] through the 
default context from the general section in sip.conf  it will generate 
the correct response, but this is not because asterisk recognized the 
situation and behaved appropriately, but because it's broken and doesn't 
check for the correct context to determine if the call should return a 
200/OK or whether it should say 404 because the call could never succeed 
if it where an INVITE.
This is also closely related to Asterisk SIP's lack of proper [user 
section] authentication/recognition for incoming calls. We've seen a lot 
of posts here where new users have problems with this, but the real 
problem is usually not acknowledged.

Olle E. Johansson wrote:

> Karl Brose wrote:
>
>>   If the response to an OPTIONS is generated by a proxy server, the
>

> This is what asterisk is doing, or?

see above

> Please explain where and how you think Asterisk is not following the RFC,
> and I'll look into it.
>
> The other alternative would be to act as a UAS, but that may be 
> confusing.
> Is any phone using this for checking if an URL is busy or not?
> In dialogue or out of dialogue?
>
> Just want to know if there's anything out there to test with.
> Thank you for looking this up.
> /O

I found out through an actual server that send OPTION packets. but that 
was a while ago before I fixed my drivers.
Haven't submitted it to the bug tracker.
I posted a fix for this earlier today here for the recent cvs'es, it at 
least looks up the context to test for the extension called and returns 
200 if that extension is valid. But there should really be more 
processing for more general compliance.


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