[Asterisk-Users] a few questions about sip implementation

[John Todd]

Simon -
   There are certainly some things that are lacking with Asterisk's 
SIP implementation, some of which you may have tripped over.  Of 
course, Asterisk's SIP stack is "pragmatic" and has been developed as 
people discover needs for particular functions.  However, there are 
perhaps some solutions to your questions:

>I'm looking at RFC 3261, I think the latest SIP standard and have a few
>questions about the * sip implementation:
>
>1. 8.2.6.1 Sending a Provisional Response says that UASs SHOULD NOT issue
>a provisional response to non-INVITE requests.
>
>>From my message yesterday * appears to be sending a SIP/2.0 100 Trying to
>X-Lite's REGISTER request before sending the SIP/2.0 200 OK message.
>
>Is this correct?

Yes, that is what it is doing and and while it may not adherent to 
the exact reading of the RFC, I have seen several other proxies doing 
the same thing (examples: FWD's SIP proxy (Cisco?) does send "100 
Trying" but SER does not) so I will assume it's an awkward industry 
standard, though perhaps not exactly compliant to the RFC paragraph 
that you describe.

>2. 10.3 Processing REGISTER requests. The 5th paragraph states "that the
>registrar has to know the set of domain(s) for which it maintains
>bindings".
>
>How is this specified in Asterisk?  Through the context?  ie should the
>domains be specified in sip.conf via the context parameter, of the
>form context=domain.com?

Currently, that is not possible.  The registrar maintains the set of 
allowed bindings by an exactly matching username/password process, OR 
via an IP-based permissions scheme.  There are no wildcards for 
domains in Asterisk for authentication.  This does not preclude you 
from creating entries for each SIP user in your sip.conf file, which 
is a bit tedious, but may be an interim solution.

>3. I have another SIP account (sip:912345678 at domain.es) which I would like
>to use within asterisk both for dialing out and for receiving calls.
>
>I see that sip.conf has a line
>
>register => 912345678 at domain.es/1234
>
>where 1234 is the local asterisk extension.  From chan_sip.c, line 1390 I
>see that I can use the form:
>
>register => user[:secret[:authuser]]@host[:port][/localextension]
>
>However my registrar requires that I authenticate with domain.es, but use
>a sip proxy at ip 1.2.3.4, the two are unrelated and domain.es has no ip
>address.  How can I get Asterisk to register with the remote prxoy?

I sent a note to Mark about this.  We had discussed a patch to fix 
this back in Feb, but apparently it didn't do quite the trick (due to 
my lack of testing.)  Lines like this are supported:

register => 9993 at something.foo.net:password at blatz.filbert.com/9993

However, the current REGISTER routines don't chop off 
"blatz.filbert.com" in the To: and From: fields.  Hopefully a repair 
will be seen in the future.

JT


>Sorry for the newbie questions.
>
>Simon
>