[Asterisk-Users] SIP protocol bug ???

[mtm spm]

Hi Olle,

--- "Olle E. Johansson" <oej at edvina.net> wrote:
> The first Invite is without credentials, since
> digest authentication needs input
> from the server to create credentials.

This is also what I understood too from rfc.
I was just confused becouse in the Asterisk code
there was something like this:

case 401: /* Not authorized on REGISTER */
	if (p->registry && !strcasecmp(msg, "REGISTER")) {
		if ((p->authtries > 1) || do_register_auth(p, req,
"WWW-Authenticate", "Authorization")) {
			ast_log(LOG_NOTICE, "Failed to authenticate on
REGISTER to '%s'\n", get_header(&p->initreq, "From"));
			p->needdestroy = 1;
} else	p->needdestroy = 1;				

So, only the case of registration was handled in 401.

However I just added something like:

if(!strcasecmp(msg, "INVITE")) {
		if ((p->authtries > 1) || do_proxy_auth2(p, req,
"INVITE", 1)) {
			ast_log(LOG_NOTICE, "Failed to authenticate on
INVITE to '%s'\n", get_header(&p->initreq, "From"));
			p->needdestroy = 1;
}
and I implemented the new function do_proxy_auth2 as a
modified version of do_proxy_auth to work with
"Authorization" instead of "Proxy-Authorization" and I
got the thing to work fine.

However, I wanted to check with others who was wrong:
Asterisk or my SIP provider.
Is this the right thing to do in respect to the
standard?

Tnx,
MTM


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