ast_callerid restructuring

The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.

Eliminate struct ast_callerid and replace it with the following struct
organization:

struct ast_party_name {
	char *str;
	int char_set;
	int presentation;
	unsigned char valid;
};
struct ast_party_number {
	char *str;
	int plan;
	int presentation;
	unsigned char valid;
};
struct ast_party_subaddress {
	char *str;
	int type;
	unsigned char odd_even_indicator;
	unsigned char valid;
};
struct ast_party_id {
	struct ast_party_name name;
	struct ast_party_number number;
	struct ast_party_subaddress subaddress;
	char *tag;
};
struct ast_party_dialed {
	struct {
		char *str;
		int plan;
	} number;
	struct ast_party_subaddress subaddress;
	int transit_network_select;
};
struct ast_party_caller {
	struct ast_party_id id;
	char *ani;
	int ani2;
};

The new organization adds some new information as well.

* The party name and number now have their own presentation value that can
be manipulated independently.  ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.

* The party name and number now have a valid flag.  Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.

* The party name now has a character set value.  SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.

* The dialed party now has a numbering plan value that could be useful to
have available.

The various channel drivers will need to be updated to support the new
core features as needed.  They have simply been converted to supply
current functionality at this time.


The following items of note were either corrected or enhanced:

* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.

* CALLERPRES() is now deprecated because the name and number have their
own presentation values.

* Fixed app_alarmreceiver.c write_metadata().  The workstring[] could
contain garbage.  It also can only contain the caller id number so using
ast_callerid_parse() on it is silly.  There was also a typo in the
CALLERNAME if test.

* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string.  ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string.  Then using
ast_shrink_phone_number() could alter it even more.

* Fixed caller ID name and number memory leak in chan_usbradio.c.

* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.

* Protected access to a caller channel with lock in chan_sip.c.

* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk().  Also made save all caller ID data instead of just the name
and number strings.

* Simplified cdr.c set_one_cid().  It hand coded the ast_callerid_merge()
function.

* Corrected some weirdness with app_privacy.c's use of caller
presentation.

Review:	https://reviewboard.asterisk.org/r/702/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 199 insertions, 101 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5d6d2f198..8dc4c5dbe 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1457,7 +1457,6 @@ static int get_msg_text(char *buf, int len, struct sip_request *req, int addnewl
 static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout);
 static void update_connectedline(struct sip_pvt *p, const void *data, size_t datalen);
 static void update_redirecting(struct sip_pvt *p, const void *data, size_t datalen);
-static void change_redirecting_information(struct sip_pvt *p, struct sip_request *req, struct ast_party_redirecting *redirecting, int set_call_forward);
 static int get_domain(const char *str, char *domain, int len);
 static void get_realm(struct sip_pvt *p, const struct sip_request *req);
 
@@ -5073,7 +5072,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
 		ast->hangupcause = AST_CAUSE_USER_BUSY;
 		return res;
 	}
-	p->callingpres = ast->cid.cid_pres;
+	p->callingpres = ast_party_id_presentation(&ast->caller.id);
 	p->jointcapability = ast_rtp_instance_available_formats(p->rtp, p->capability, p->prefcodec);
 	p->jointnoncodeccapability = p->noncodeccapability;
 
@@ -6390,7 +6389,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 	ast_channel_lock(tmp);
 	sip_pvt_lock(i);
 	ast_channel_cc_params_init(tmp, i->cc_params);
-	tmp->cid.cid_tag = ast_strdup(i->cid_tag);
+	tmp->caller.id.tag = ast_strdup(i->cid_tag);
 	ast_channel_unlock(tmp);
 
 	tmp->tech = ( ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INFO || ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_SHORTINFO) ?  &sip_tech_info : &sip_tech;
@@ -6491,7 +6490,8 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 
 	tmp->callgroup = i->callgroup;
 	tmp->pickupgroup = i->pickupgroup;
-	tmp->cid.cid_pres = i->callingpres;
+	tmp->caller.id.name.presentation = i->callingpres;
+	tmp->caller.id.number.presentation = i->callingpres;
 	if (!ast_strlen_zero(i->parkinglot))
 		ast_string_field_set(tmp, parkinglot, i->parkinglot);
 	if (!ast_strlen_zero(i->accountcode))
@@ -6513,12 +6513,15 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
 
 	/* Don't use ast_set_callerid() here because it will
 	 * generate an unnecessary NewCallerID event  */
-	tmp->cid.cid_ani = ast_strdup(i->cid_num);
-	if (!ast_strlen_zero(i->rdnis))
-		tmp->redirecting.from.number = ast_strdup(i->rdnis);
-	
-	if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s"))
-		tmp->cid.cid_dnid = ast_strdup(i->exten);
+	tmp->caller.ani = ast_strdup(i->cid_num);
+	if (!ast_strlen_zero(i->rdnis)) {
+		tmp->redirecting.from.number.valid = 1;
+		tmp->redirecting.from.number.str = ast_strdup(i->rdnis);
+	}
+
+	if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s")) {
+		tmp->dialed.number.str = ast_strdup(i->exten);
+	}
 
 	tmp->priority = 1;
 	if (!ast_strlen_zero(i->uri))
@@ -6817,7 +6820,8 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
 		if (strcmp(ast->exten, "fax")) {
 			const char *target_context = S_OR(ast->macrocontext, ast->context);
 			ast_channel_unlock(ast);
-			if (ast_exists_extension(ast, target_context, "fax", 1, ast->cid.cid_num)) {
+			if (ast_exists_extension(ast, target_context, "fax", 1,
+				S_COR(ast->caller.id.number.valid, ast->caller.id.number.str, NULL))) {
 				ast_verbose(VERBOSE_PREFIX_2 "Redirecting '%s' to fax extension due to CNG detection\n", ast->name);
 				pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast->exten);
 				if (ast_async_goto(ast, target_context, "fax", 1)) {
@@ -6826,11 +6830,11 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
 				fr = &ast_null_frame;
 			} else {
 				ast_log(LOG_NOTICE, "FAX CNG detected but no fax extension\n");
-                        }
+			}
 		} else {
 			ast_channel_unlock(ast);
-                }
-        }
+		}
+	}
 
 	/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
 	if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
@@ -8228,7 +8232,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 					if (strcmp(p->owner->exten, "fax")) {
 						const char *target_context = S_OR(p->owner->macrocontext, p->owner->context);
 						ast_channel_unlock(p->owner);
-						if (ast_exists_extension(p->owner, target_context, "fax", 1, p->owner->cid.cid_num)) {
+						if (ast_exists_extension(p->owner, target_context, "fax", 1,
+							S_COR(p->owner->caller.id.number.valid, p->owner->caller.id.number.str, NULL))) {
 							ast_verbose(VERBOSE_PREFIX_2 "Redirecting '%s' to fax extension due to peer T.38 re-INVITE\n", p->owner->name);
 							pbx_builtin_setvar_helper(p->owner, "FAXEXTEN", p->owner->exten);
 							if (ast_async_goto(p->owner, target_context, "fax", 1)) {
@@ -9691,11 +9696,15 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p)
 		return 0;
 	}
 
-	if (p->owner && p->owner->connected.id.number)
-		lid_num = p->owner->connected.id.number;
-	if (p->owner && p->owner->connected.id.name)
-		lid_name = p->owner->connected.id.name;
-	lid_pres = (p->owner) ? p->owner->connected.id.number_presentation : AST_PRES_NUMBER_NOT_AVAILABLE;
+	if (p->owner && p->owner->connected.id.number.valid
+		&& p->owner->connected.id.number.str) {
+		lid_num = p->owner->connected.id.number.str;
+	}
+	if (p->owner && p->owner->connected.id.name.valid
+		&& p->owner->connected.id.name.str) {
+		lid_name = p->owner->connected.id.name.str;
+	}
+	lid_pres = (p->owner) ? ast_party_id_presentation(&p->owner->connected.id) : AST_PRES_NUMBER_NOT_AVAILABLE;
 
 	if (ast_strlen_zero(lid_num))
 		return 0;
@@ -10700,14 +10709,16 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
 	snprintf(p->lastmsg, sizeof(p->lastmsg), "Init: %s", sip_methods[sipmethod].text);
 
 	d = S_OR(p->fromdomain, ast_sockaddr_stringify_host(&p->ourip));
-	if (p->owner && (p->owner->connected.id.number_presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
-		l = p->owner->connected.id.number;
-		n = p->owner->connected.id.name;
-	} else if (p->owner && (p->owner->connected.id.number_presentation & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED && (!ast_test_flag(&p->flags[0], SIP_SENDRPID))) {
-		/* if we are not sending RPID and user wants his callerid restricted */		
-		l = CALLERID_UNKNOWN;
-		n = l;
-		d = FROMDOMAIN_INVALID;
+	if (p->owner) {
+		if ((ast_party_id_presentation(&p->owner->connected.id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
+			l = p->owner->connected.id.number.valid ? p->owner->connected.id.number.str : NULL;
+			n = p->owner->connected.id.name.valid ? p->owner->connected.id.name.str : NULL;
+		} else if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
+			/* if we are not sending RPID and user wants his callerid restricted */		
+			l = CALLERID_UNKNOWN;
+			n = l;
+			d = FROMDOMAIN_INVALID;
+		}
 	}
 
 	/* Hey, it's a NOTIFY! See if they've configured a mwi_from.
@@ -10845,16 +10856,18 @@ static void add_diversion_header(struct sip_request *req, struct sip_pvt *pvt)
 		return;
 	}
 
-	diverting_number = pvt->owner->redirecting.from.number;
-	diverting_name = pvt->owner->redirecting.from.name;
-	reason = sip_reason_code_to_str(pvt->owner->redirecting.reason);
-
-	if (ast_strlen_zero(diverting_number)) {
+	diverting_number = pvt->owner->redirecting.from.number.str;
+	if (!pvt->owner->redirecting.from.number.valid
+		|| ast_strlen_zero(diverting_number)) {
 		return;
 	}
 
+	reason = sip_reason_code_to_str(pvt->owner->redirecting.reason);
+
 	/* We at least have a number to place in the Diversion header, which is enough */
-	if (ast_strlen_zero(diverting_name)) {
+	diverting_name = pvt->owner->redirecting.from.name.str;
+	if (!pvt->owner->redirecting.from.name.valid
+		|| ast_strlen_zero(diverting_name)) {
 		snprintf(header_text, sizeof(header_text), "<sip:%s@%s>;reason=%s", diverting_number,
 				ast_sockaddr_stringify_host(&pvt->ourip), reason);
 	} else {
@@ -11291,11 +11304,17 @@ static void state_notify_build_xml(int state, int full, const char *exten, const
 				struct ast_channel *caller;
 
 				if ((caller = ast_channel_callback(find_calling_channel, NULL, p, 0))) {
-					int need = strlen(caller->cid.cid_num) + strlen(p->fromdomain) + sizeof("sip:@");
-					local_target = alloca(need);
+					char *cid_num;
+					int need;
+
 					ast_channel_lock(caller);
-					snprintf(local_target, need, "sip:%s@%s", caller->cid.cid_num, p->fromdomain);
-					local_display = ast_strdupa(caller->cid.cid_name);
+					cid_num = S_COR(caller->caller.id.number.valid,
+						caller->caller.id.number.str, "");
+					need = strlen(cid_num) + strlen(p->fromdomain) + sizeof("sip:@");
+					local_target = alloca(need);
+					snprintf(local_target, need, "sip:%s@%s", cid_num, p->fromdomain);
+					local_display = ast_strdupa(S_COR(caller->caller.id.name.valid,
+						caller->caller.id.name.str, ""));
 					ast_channel_unlock(caller);
 					caller = ast_channel_unref(caller);
 				}
@@ -11599,10 +11618,15 @@ static void update_connectedline(struct sip_pvt *p, const void *data, size_t dat
 
 	if (!ast_test_flag(&p->flags[0], SIP_SENDRPID))
 		return;
-	if (ast_strlen_zero(p->owner->connected.id.number))
+	if (!p->owner->connected.id.number.valid
+		|| ast_strlen_zero(p->owner->connected.id.number.str)) {
 		return;
+	}
 
-	append_history(p, "ConnectedLine", "%s party is now %s <%s>", ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "Calling" : "Called", p->owner->connected.id.name, p->owner->connected.id.number);
+	append_history(p, "ConnectedLine", "%s party is now %s <%s>",
+		ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "Calling" : "Called",
+		S_COR(p->owner->connected.id.name.valid, p->owner->connected.id.name.str, ""),
+		S_COR(p->owner->connected.id.number.valid, p->owner->connected.id.number.str, ""));
 
 	if (p->owner->_state == AST_STATE_UP || ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
 		struct sip_request req;
@@ -13551,7 +13575,8 @@ static int get_pai(struct sip_pvt *p, struct sip_request *req)
 
 	if (p->owner) {
 		ast_set_callerid(p->owner, cid_num, cid_name, NULL);
-		p->owner->cid.cid_pres = callingpres;
+		p->owner->caller.id.name.presentation = callingpres;
+		p->owner->caller.id.number.presentation = callingpres;
 	}
 
 	return 1;
@@ -13650,7 +13675,8 @@ static int get_rpid(struct sip_pvt *p, struct sip_request *oreq)
 
 	if (p->owner) {
 		ast_set_callerid(p->owner, cid_num, cid_name, NULL);
-		p->owner->cid.cid_pres = callingpres;
+		p->owner->caller.id.name.presentation = callingpres;
+		p->owner->caller.id.number.presentation = callingpres;
 	}
 
 	return 1;
@@ -17811,7 +17837,9 @@ static struct ast_custom_function sipchaninfo_function = {
 /*! \brief update redirecting information for a channel based on headers
  *
  */
-static void change_redirecting_information(struct sip_pvt *p, struct sip_request *req, struct ast_party_redirecting *redirecting, int set_call_forward)
+static void change_redirecting_information(struct sip_pvt *p, struct sip_request *req,
+	struct ast_party_redirecting *redirecting,
+	struct ast_set_party_redirecting *update_redirecting, int set_call_forward)
 {
 	char *redirecting_from_name = NULL;
 	char *redirecting_from_number = NULL;
@@ -17841,42 +17869,38 @@ static void change_redirecting_information(struct sip_pvt *p, struct sip_request
 	}
 
 	if (!ast_strlen_zero(redirecting_from_number)) {
-		if (redirecting->from.number) {
-			ast_free(redirecting->from.number);
-		}
 		ast_debug(3, "Got redirecting from number %s\n", redirecting_from_number);
-		redirecting->from.number = redirecting_from_number;
+		update_redirecting->from.number = 1;
+		redirecting->from.number.valid = 1;
+		ast_free(redirecting->from.number.str);
+		redirecting->from.number.str = redirecting_from_number;
 	}
 	if (!ast_strlen_zero(redirecting_from_name)) {
-		if (redirecting->from.name) {
-			ast_free(redirecting->from.name);
-		}
 		ast_debug(3, "Got redirecting from name %s\n", redirecting_from_name);
-		redirecting->from.name = redirecting_from_name;
+		update_redirecting->from.name = 1;
+		redirecting->from.name.valid = 1;
+		ast_free(redirecting->from.name.str);
+		redirecting->from.name.str = redirecting_from_name;
 	}
 	if (!ast_strlen_zero(p->cid_tag)) {
-		if (redirecting->from.tag) {
-			ast_free(redirecting->from.tag);
-		}
+		ast_free(redirecting->from.tag);
 		redirecting->from.tag = ast_strdup(p->cid_tag);
-		if (redirecting->to.tag) {
-			ast_free(redirecting->to.tag);
-		}
+		ast_free(redirecting->to.tag);
 		redirecting->to.tag = ast_strdup(p->cid_tag);
 	}
 	if (!ast_strlen_zero(redirecting_to_number)) {
-		if (redirecting->to.number) {
-			ast_free(redirecting->to.number);
-		}
 		ast_debug(3, "Got redirecting to number %s\n", redirecting_to_number);
-		redirecting->to.number = redirecting_to_number;
+		update_redirecting->to.number = 1;
+		redirecting->to.number.valid = 1;
+		ast_free(redirecting->to.number.str);
+		redirecting->to.number.str = redirecting_to_number;
 	}
 	if (!ast_strlen_zero(redirecting_to_name)) {
-		if (redirecting->to.name) {
-			ast_free(redirecting->to.name);
-		}
 		ast_debug(3, "Got redirecting to name %s\n", redirecting_from_number);
-		redirecting->to.name = redirecting_to_name;
+		update_redirecting->to.name = 1;
+		redirecting->to.name.valid = 1;
+		ast_free(redirecting->to.name.str);
+		redirecting->to.name.str = redirecting_to_name;
 	}
 	redirecting->reason = reason;
 }
@@ -18169,6 +18193,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 	char *p_hdrval;
 	int rtn;
 	struct ast_party_connected_line connected;
+	struct ast_set_party_connected_line update_connected;
 
 	if (reinvite)
 		ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
@@ -18224,12 +18249,23 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 		if (!req->ignore && p->owner) {
 			if (get_rpid(p, req)) {
 				ast_party_connected_line_init(&connected);
-				connected.id.number = (char *) p->cid_num;
-				connected.id.name = (char *) p->cid_name;
+				memset(&update_connected, 0, sizeof(update_connected));
+				if (p->cid_num) {
+					update_connected.id.number = 1;
+					connected.id.number.valid = 1;
+					connected.id.number.str = (char *) p->cid_num;
+					connected.id.number.presentation = p->callingpres;
+				}
+				if (p->cid_name) {
+					update_connected.id.name = 1;
+					connected.id.name.valid = 1;
+					connected.id.name.str = (char *) p->cid_name;
+					connected.id.name.presentation = p->callingpres;
+				}
 				connected.id.tag = (char *) p->cid_tag;
-				connected.id.number_presentation = p->callingpres;
 				connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
-				ast_channel_queue_connected_line_update(p->owner, &connected);
+				ast_channel_queue_connected_line_update(p->owner, &connected,
+					&update_connected);
 			}
 			sip_handle_cc(p, req, AST_CC_CCNR);
 			ast_queue_control(p->owner, AST_CONTROL_RINGING);
@@ -18254,9 +18290,15 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 		if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
 			ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
 		if (!req->ignore && p->owner) {
-			struct ast_party_redirecting redirecting = {{0,},};
-			change_redirecting_information(p, req, &redirecting, FALSE);
-			ast_channel_queue_redirecting_update(p->owner, &redirecting);
+			struct ast_party_redirecting redirecting;
+			struct ast_set_party_redirecting update_redirecting;
+
+			ast_party_redirecting_init(&redirecting);
+			memset(&update_redirecting, 0, sizeof(update_redirecting));
+			change_redirecting_information(p, req, &redirecting, &update_redirecting,
+				FALSE);
+			ast_channel_queue_redirecting_update(p->owner, &redirecting,
+				&update_redirecting);
 			ast_party_redirecting_free(&redirecting);
 			sip_handle_cc(p, req, AST_CC_CCNR);
 		}
@@ -18270,12 +18312,23 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 			if (get_rpid(p, req)) {
 				/* Queue a connected line update */
 				ast_party_connected_line_init(&connected);
-				connected.id.number = (char *) p->cid_num;
-				connected.id.name = (char *) p->cid_name;
+				memset(&update_connected, 0, sizeof(update_connected));
+				if (p->cid_num) {
+					update_connected.id.number = 1;
+					connected.id.number.valid = 1;
+					connected.id.number.str = (char *) p->cid_num;
+					connected.id.number.presentation = p->callingpres;
+				}
+				if (p->cid_name) {
+					update_connected.id.name = 1;
+					connected.id.name.valid = 1;
+					connected.id.name.str = (char *) p->cid_name;
+					connected.id.name.presentation = p->callingpres;
+				}
 				connected.id.tag = (char *) p->cid_tag;
-				connected.id.number_presentation = p->callingpres;
 				connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
-				ast_channel_queue_connected_line_update(p->owner, &connected);
+				ast_channel_queue_connected_line_update(p->owner, &connected,
+					&update_connected);
 			}
 			sip_handle_cc(p, req, AST_CC_CCNR);
 		}
@@ -18316,12 +18369,23 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 		if (!req->ignore && p->owner && (get_rpid(p, req) || !reinvite)) {
 			/* Queue a connected line update */
 			ast_party_connected_line_init(&connected);
-			connected.id.number = (char *) p->cid_num;
-			connected.id.name = (char *) p->cid_name;
+			memset(&update_connected, 0, sizeof(update_connected));
+			if (p->cid_num) {
+				update_connected.id.number = 1;
+				connected.id.number.valid = 1;
+				connected.id.number.str = (char *) p->cid_num;
+				connected.id.number.presentation = p->callingpres;
+			}
+			if (p->cid_name) {
+				update_connected.id.name = 1;
+				connected.id.name.valid = 1;
+				connected.id.name.str = (char *) p->cid_name;
+				connected.id.name.presentation = p->callingpres;
+			}
 			connected.id.tag = (char *) p->cid_tag;
-			connected.id.number_presentation = p->callingpres;
 			connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
-			ast_channel_queue_connected_line_update(p->owner, &connected);
+			ast_channel_queue_connected_line_update(p->owner, &connected,
+				&update_connected);
 		}
 
 		/* Parse contact header for continued conversation */
@@ -19234,9 +19298,14 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
 				case 302: /* Moved temporarily */
 				case 305: /* Use Proxy */
 				if (p->owner) {
-					struct ast_party_redirecting redirecting = {{0,},};
-					change_redirecting_information(p, req, &redirecting, TRUE);
-					ast_channel_set_redirecting(p->owner, &redirecting);
+					struct ast_party_redirecting redirecting;
+					struct ast_set_party_redirecting update_redirecting;
+
+					ast_party_redirecting_init(&redirecting);
+					change_redirecting_information(p, req, &redirecting,
+						&update_redirecting, TRUE);
+					ast_channel_set_redirecting(p->owner, &redirecting,
+						&update_redirecting);
 					ast_party_redirecting_free(&redirecting);
 				}
 					/* Fall through */
@@ -20454,13 +20523,24 @@ static int handle_request_update(struct sip_pvt *p, struct sip_request *req)
 	}
 	if (get_rpid(p, req)) {
 		struct ast_party_connected_line connected;
+		struct ast_set_party_connected_line update_connected;
 		ast_party_connected_line_init(&connected);
-		connected.id.number = (char *) p->cid_num;
-		connected.id.name = (char *) p->cid_name;
+		memset(&update_connected, 0, sizeof(update_connected));
+		if (p->cid_num) {
+			update_connected.id.number = 1;
+			connected.id.number.valid = 1;
+			connected.id.number.str = (char *) p->cid_num;
+			connected.id.number.presentation = p->callingpres;
+		}
+		if (p->cid_name) {
+			update_connected.id.name = 1;
+			connected.id.name.valid = 1;
+			connected.id.name.str = (char *) p->cid_name;
+			connected.id.name.presentation = p->callingpres;
+		}
 		connected.id.tag = (char *) p->cid_tag;
-		connected.id.number_presentation = p->callingpres;
 		connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
-		ast_channel_queue_connected_line_update(p->owner, &connected);
+		ast_channel_queue_connected_line_update(p->owner, &connected, &update_connected);
 	}
 	transmit_response(p, "200 OK", req);
 	return 0;
@@ -20486,6 +20566,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 	struct sip_peer *authpeer = NULL;	/* Matching Peer */
 	int reinvite = 0;
 	int rtn;
+	struct ast_party_redirecting redirecting;
+	struct ast_set_party_redirecting update_redirecting;
 
 	const char *p_uac_se_hdr;       /* UAC's Session-Expires header string                      */
 	const char *p_uac_min_se;       /* UAC's requested Min-SE interval (char string)            */
@@ -20783,14 +20865,26 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 			ast_clear_flag(&p->flags[0], SIP_OUTGOING);	/* This is now an inbound dialog */
 			if (get_rpid(p, req)) {
 				struct ast_party_connected_line connected;
+				struct ast_set_party_connected_line update_connected;
 
 				ast_party_connected_line_init(&connected);
-				connected.id.number = (char *) p->cid_num;
-				connected.id.name = (char *) p->cid_name;
+				memset(&update_connected, 0, sizeof(update_connected));
+				if (p->cid_num) {
+					update_connected.id.number = 1;
+					connected.id.number.valid = 1;
+					connected.id.number.str = (char *) p->cid_num;
+					connected.id.number.presentation = p->callingpres;
+				}
+				if (p->cid_name) {
+					update_connected.id.name = 1;
+					connected.id.name.valid = 1;
+					connected.id.name.str = (char *) p->cid_name;
+					connected.id.name.presentation = p->callingpres;
+				}
 				connected.id.tag = (char *) p->cid_tag;
-				connected.id.number_presentation = p->callingpres;
 				connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
-				ast_channel_queue_connected_line_update(p->owner, &connected);
+				ast_channel_queue_connected_line_update(p->owner, &connected,
+					&update_connected);
 			}
 			/* Handle SDP here if we already have an owner */
 			if (find_sdp(req)) {
@@ -20958,16 +21052,19 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 			build_route(p, req, 0);
 
 			if (c) {
-				struct ast_party_redirecting redirecting = { { 0, }, };
+				ast_party_redirecting_init(&redirecting);
+				memset(&update_redirecting, 0, sizeof(update_redirecting));
 				/* Pre-lock the call */
 				ast_channel_lock(c);
-				change_redirecting_information(p, req, &redirecting, FALSE); /*Will return immediately if no Diversion header is present */
-				ast_channel_set_redirecting(c, &redirecting);
+				change_redirecting_information(p, req, &redirecting, &update_redirecting,
+					FALSE); /*Will return immediately if no Diversion header is present */
+				ast_channel_set_redirecting(c, &redirecting, &update_redirecting);
 				ast_party_redirecting_free(&redirecting);
 			}
 		}
 	} else {
-		struct ast_party_redirecting redirecting = {{0,},};
+		ast_party_redirecting_init(&redirecting);
+		memset(&update_redirecting, 0, sizeof(update_redirecting));
 		if (sipdebug) {
 			if (!req->ignore)
 				ast_debug(2, "Got a SIP re-invite for call %s\n", p->callid);
@@ -20977,9 +21074,9 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 		if (!req->ignore)
 			reinvite = 1;
 		c = p->owner;
-		change_redirecting_information(p, req, &redirecting, FALSE); /*Will return immediately if no Diversion header is present */
+		change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE); /*Will return immediately if no Diversion header is present */
 		if (c) {
-			ast_channel_set_redirecting(c, &redirecting);
+			ast_channel_set_redirecting(c, &redirecting, &update_redirecting);
 		}
 		ast_party_redirecting_free(&redirecting);
 	}
@@ -21448,8 +21545,8 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
 		ast_indicate(target.chan1, AST_CONTROL_UNHOLD);
 
 		if (target.chan2) {
-			ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee);
-			ast_channel_queue_connected_line_update(target.chan2, &connected_to_target);
+			ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee, NULL);
+			ast_channel_queue_connected_line_update(target.chan2, &connected_to_target, NULL);
 		} else {
 			/* Since target.chan1 isn't actually connected to another channel, there is no way for us
 			 * to queue a frame so that its connected line status will be updated.
@@ -21462,7 +21559,8 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
 			int payload_size;
 			int frame_size;
 			unsigned char connected_line_data[1024];
-			payload_size = ast_connected_line_build_data(connected_line_data, sizeof(connected_line_data), &connected_to_target);
+			payload_size = ast_connected_line_build_data(connected_line_data,
+				sizeof(connected_line_data), &connected_to_target, NULL);
 			frame_size = payload_size + sizeof(*frame_payload);
 			if (payload_size != -1 && (frame_payload = alloca(frame_size))) {
 				frame_payload->payload_size = payload_size;
@@ -21474,7 +21572,7 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
 			 * will be updated, we also are going to queue a plain old connected line update on target.chan1. This
 			 * way, either Dial or Queue can apply this connected line update to the outgoing ringing channel.
 			 */
-			ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee);
+			ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee, NULL);
 
 		}
 		ast_channel_unref(current->chan1);