[asterisk-commits] jrose: branch 1.8 r371357 - /branches/1.8/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: jrose
Date: Thu Aug 16 13:57:27 2012
New Revision: 371357

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=371357
Log:
chan_sip: Use pvt outgoing_call variable to set Remote-Party-ID Header

Previously the pvt SIP_OUTGOING flag was used instead, which will frequently
flip during reinvites.

(closes issue AST-897)
Reported by: Thomas Arimont

Modified:
    branches/1.8/channels/chan_sip.c

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=371357&r1=371356&r2=371357
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Thu Aug 16 13:57:27 2012
@@ -10999,7 +10999,7 @@
 		}
 		add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
 	} else {
-		ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>;party=%s", lid_name, lid_num, fromdomain, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "calling" : "called");
+		ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>;party=%s", lid_name, lid_num, fromdomain, p->outgoing_call ? "calling" : "called");
 
 		switch (lid_pres) {
 		case AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED: