[asterisk-commits] jrose: branch 1.8 r411189 - in /branches/1.8: channels/ configs/

[SVN commits to the Asterisk project]

Author: jrose
Date: Wed Mar 26 10:50:48 2014
New Revision: 411189

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=411189
Log:
chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)

Prior too this patch, the P-Asserted-Identity header would include anonymous
caller id information which seems to go against the point of the
P-Asserted-Identity header. Now the real caller ID information will be
included in this header. Also, no privacy header would be included.
This patch adds 'Privacy: id' to outgoing SIP messages that include the
P-Asserted-Identity header.

(closes issue AST-1301)


Modified:
    branches/1.8/channels/chan_sip.c
    branches/1.8/configs/sip.conf.sample

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=411189&r1=411188&r2=411189
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Wed Mar 26 10:50:48 2014
@@ -11406,7 +11406,6 @@
 	const char *fromdomain;
 	const char *privacy = NULL;
 	const char *screen = NULL;
-	const char *anonymous_string = "\"Anonymous\" <sip:anonymous at anonymous.invalid>";
 
 	if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
 		return 0;
@@ -11434,12 +11433,11 @@
 	lid_num = ast_uri_encode(lid_num, tmp2, sizeof(tmp2), 0);
 
 	if (ast_test_flag(&p->flags[0], SIP_SENDRPID_PAI)) {
+		ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
+		add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
 		if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
-			ast_str_set(&tmp, -1, "%s", anonymous_string);
-		} else {
-			ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
-		}
-		add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
+			add_header(req, "Privacy", "id");
+		}
 	} else {
 		ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>;party=%s", lid_name, lid_num, fromdomain, p->outgoing_call ? "calling" : "called");
 

Modified: branches/1.8/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/configs/sip.conf.sample?view=diff&rev=411189&r1=411188&r2=411189
==============================================================================
--- branches/1.8/configs/sip.conf.sample (original)
+++ branches/1.8/configs/sip.conf.sample Wed Mar 26 10:50:48 2014
@@ -1301,7 +1301,8 @@
 ;allow=g723.1                    ; Asterisk only supports g723.1 pass-thru!
 ;allow=g729                      ; Pass-thru only unless g729 license obtained
 ;callingpres=allowed_passed_screen ; Set caller ID presentation
-                                 ; See README.callingpres for more information
+                                 ; See function CALLERPRES documentation for possible
+                                 ; values.
 
 ;[xlite1]
 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!