From: Jonathan Rose <jrose@digium.com>
Date: Tue, 15 Apr 2014 15:21:27 +0000 (+0000)
Subject: Reverting r411189 so that it can be put up for public review
X-Git-Tag: 1.8.28.0-rc1~32
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=445b1ced10ebcb465f7b4bb4863569e00f126581;p=thirdparty%2Fasterisk.git

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)
---



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 971fb095f3..3b961a869b 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -11394,6 +11394,7 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p)
 	const char *fromdomain;
 	const char *privacy = NULL;
 	const char *screen = NULL;
+	const char *anonymous_string = "\"Anonymous\" <sip:anonymous@anonymous.invalid>";
 
 	if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
 		return 0;
@@ -11421,11 +11422,12 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p)
 	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) {
-			add_header(req, "Privacy", "id");
+			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));
 	} else {
 		ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>;party=%s", lid_name, lid_num, fromdomain, p->outgoing_call ? "calling" : "called");
 
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 3e1cb08a8d..cd12029bdc 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -1301,8 +1301,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;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)!