From fa3a2f8ecaeb5ba65ff2ff99097aa4381fbc1379 Mon Sep 17 00:00:00 2001
From: Jonathan Rose <jrose@digium.com>
Date: Wed, 26 Mar 2014 16:15:12 +0000
Subject: [PATCH] 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)
........
Merged revisions 411189 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 411190 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 411193 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411194 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 8 +++-----
configs/sip.conf.sample | 3 ++-
2 files changed, 5 insertions(+), 6 deletions(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index dbd3903a34..231f376a8d 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12645,7 +12645,6 @@ 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>";
struct ast_party_id connected_id;
if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
@@ -12671,12 +12670,11 @@ 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) {
- 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, "Privacy", "id");
}
- 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 46af790434..2b26589d90 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -1431,7 +1431,8 @@ 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 README.callingpres for more information
+ ; See function CALLERPRES documentation for possible
+ ; values.
;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
--
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