[Asterisk-Dev] Changes to SIP

[asterisk at jdennis.net]

I'm not sure how many people are using the SIP stuff but I have noticed that
a few changes need to be made for asterisk to correctly work with a service
provider running Cisco proxy. I am not very familiar with SIP and basically
would like comments on these points before attempting to make modifications.
I do not wish to break it for anything else:

(o)  The From header needs to be set to the account that performs the
REGISTER. I currently have a small hack that allows me to specify what the
 From header should be in the sip.conf file for that peer.

(o)  When a BYE is sent the authorization digest is not sent, cisco replies
with Proxy auth required but by then Asterisk has already destroyed the
channel. From a glance it looks like the cisco phones send the same auth
digest as what was sent for the INVITE. The phone does not go through the
whole transaction of trying without auth, then receiving a proxy-auth
required error, then finally sending the auth digest.

(o)  While an INVITE is "Trying" a CANCEL request sent by asterisk results
in a "487 Request Cancelled". Cisco equipment ACKs this response. It seems
that if the proxy does not receive this ACK it does not hang up the other
end until a time out period is reached. Should Asterisk be modified to not
clean up after a CANCEL and only after sending an ACK to a 487? Does
everything always send a 487 or should perhaps a channel timeout if non is
received?

(o)  The ACK sent on the INVITE's 200 OK are not recognized for some reason.
This may be due to my from header hack I'm not sure. After each ACK sent the
proxy sends another 200 OK. Eventually the proxy drops the connection.

All experiences and ideas welcome! :)

James