[svn-commits] jrose: branch 12 r412330 - in /branches/12: ./ channels/ configs/

[SVN commits to the Digium repositories]

Author: jrose
Date: Tue Apr 15 10:58:42 2014
New Revision: 412330

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=412330
Log:
Reverting r411189 so that it can be put up for public review

---
  r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines

  chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)

  Prior to 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)
---
........

Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 412329 from http://svn.asterisk.org/svn/asterisk/branches/11

Modified:
    branches/12/   (props changed)
    branches/12/channels/chan_sip.c
    branches/12/configs/sip.conf.sample

Propchange: branches/12/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.

Modified: branches/12/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_sip.c?view=diff&rev=412330&r1=412329&r2=412330
==============================================================================
--- branches/12/channels/chan_sip.c (original)
+++ branches/12/channels/chan_sip.c Tue Apr 15 10:58:42 2014
@@ -12653,6 +12653,7 @@
 	const char *privacy = NULL;
 	const char *screen = NULL;
 	struct ast_party_id connected_id;
+	const char *anonymous_string = "\"Anonymous\" <sip:anonymous at anonymous.invalid>";
 
 	if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
 		return 0;
@@ -12677,11 +12678,12 @@
 	lid_num = ast_uri_encode(lid_num, tmp2, sizeof(tmp2), ast_uri_sip_user);
 
 	if (ast_test_flag(&p->flags[0], SIP_SENDRPID_PAI)) {
-		ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
+		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));
-		if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
-			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/12/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/12/configs/sip.conf.sample?view=diff&rev=412330&r1=412329&r2=412330
==============================================================================
--- branches/12/configs/sip.conf.sample (original)
+++ branches/12/configs/sip.conf.sample Tue Apr 15 10:58:42 2014
@@ -1431,8 +1431,7 @@
 ;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 function CALLERPRES documentation for possible
-                                 ; values.
+                                 ; See README.callingpres for more information
 
 ;[xlite1]
 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!