From cc4a0a7fc9f2154e47f871650a2ec52c8c5bafc6 Mon Sep 17 00:00:00 2001
From: Jonathan Rose <jrose@digium.com>
Date: Tue, 15 Apr 2014 16:13:35 +0000
Subject: [PATCH] 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
........
Merged revisions 412330 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 8 +++++---
configs/sip.conf.sample | 3 +--
2 files changed, 6 insertions(+), 5 deletions(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 24a9370a71..cf71ac4532 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12631,6 +12631,7 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p)
const char *privacy = NULL;
const char *screen = NULL;
struct ast_party_id connected_id;
+ const char *anonymous_string = "\"Anonymous\" <sip:anonymous@anonymous.invalid>";
if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
return 0;
@@ -12655,11 +12656,12 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p)
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);
- 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 2b26589d90..46af790434 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -1431,8 +1431,7 @@ 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)!
--
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