[asterisk-commits] rizzo: branch rizzo/astobj2 r51229 - /team/rizzo/astobj2/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: rizzo
Date: Thu Jan 18 10:40:41 2007
New Revision: 51229

URL: http://svn.digium.com/view/asterisk?view=rev&rev=51229
Log:
merge from trunk 48979,
reset invitestate when sending new invite.
Trunk does it slightly differently, i am not sure
which one is correct so i will keep my way until i
understand it better.


Modified:
    team/rizzo/astobj2/channels/chan_sip.c

Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=51229&r1=51228&r2=51229
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Thu Jan 18 10:40:41 2007
@@ -12367,6 +12367,9 @@
 		/* Then we AUTH */
 		ast_string_field_free(p, theirtag);	/* forget their old tag, so we don't match tags when getting response */
 		if (!req_ignore(req)) {
+			/* XXX trunk has if (p->authtries < MAX_AUTHTRIES) ...
+			 * but i am not sure if it is correct.
+			 */
 			p->invitestate = INV_CALLING;	/* XXX reset state */
 			if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
 				ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));