[Asterisk-Users] SIP changes in CVS head

[Olle E. Johansson]

In the development tree of Asterisk we've changed two things lately that
may affect you:

* Asterisk can fail an outbound registration

If you enter a register= statement with an incorrect password, wrong
hostname or anything else that is wrong, Asterisk will give up
registration after 10 attempts. If you do "sip show registry" you will
see the registration status as "failed". Doing a "sip reload" will
restart the attempts after you fixed the problem.

In earlier versions, Asterisk just kept sending packets endlessly.

* Authentication changes
Asterisk will now check that the authorisation user name (digest
username=) in an incoming authentication is the same as the username
part of the From: SIP uri. We've always based peer/user matching on the
username part and now we will send an error if the authentication
username is different from the From: username.

We have also fixed a problem that is occuring with Sipura devices,
where the Sipura sends a proper authentication based on an old nonce.

* Coming changes:
Changes that are in the bug tracker and hopefully will be implemented soon:
- SIP domain support: As an option, you can configure your Asterisk to
be SIP domain aware. If Asterisk gets messages directed to domains that
are not configured as local domains, the message will be rejected. Very
much like a mail server that doesn't forward or handle mail to other
domains than local domains.
This feature also makes it possible to implement SIP transfers in a
correct way, since Asterisk is able to judge whether a transfer is to a
local or remote extension.

- Support for supported/required headers: If a SIP service requires
support of a SIP extension by using the Require: header, Asterisk
doesn't understand that today. We should fail the transaction if we do
not support the extension (which is simple, since Asterisk does not
support any extensions at all today). This will also help implementing
support for the Replaces: extension in SIP transfers.

- Support of SIP call timers: This is something I just started
exploring, but we need to implement. There are a lot of reports open
in the bug tracker where we do not handle retransmissions properly. In
some cases, this is because we retransmit too often according to the SIP
standard, so we're getting totally out of sync with other devices.

The SIPURA bug fix will be implemented in the release ("stable") version
of Asterisk if Russell approves, but the others are new functions that
will only be implemented in the development version.

Meet me in Madrid to discuss chan_sip :-)

/Olle

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