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)
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Merged revisions 411189 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 411190 from http://svn.asterisk.org/svn/asterisk/branches/11
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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

@@ -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");
 

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)!