This commit introduces COLP/CONP and Redirecting party information into Asterisk.

The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.

A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.

This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.

(closes issue #8824)
Reported by: gareth

Review: http://reviewboard.digium.com/r/201



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 60 insertions, 1 deletions

diff --git a/main/features.c b/main/features.c
index 0b6d7f53c..8efe0dbf7 100644
--- a/main/features.c
+++ b/main/features.c
@@ -1387,6 +1387,7 @@ static int builtin_atxfer(struct ast_channel *chan, struct ast_channel *peer, st
 	struct ast_bridge_config bconfig;
 	struct ast_frame *f;
 	int l;
+	struct ast_party_connected_line connected_line = {{0,},};
 	struct ast_datastore *features_datastore;
 	struct ast_dial_features *dialfeatures = NULL;
 
@@ -1478,6 +1479,17 @@ static int builtin_atxfer(struct ast_channel *chan, struct ast_channel *peer, st
 		memset(&bconfig,0,sizeof(struct ast_bridge_config));
 		ast_set_flag(&(bconfig.features_caller), AST_FEATURE_DISCONNECT);
 		ast_set_flag(&(bconfig.features_callee), AST_FEATURE_DISCONNECT);
+		/* We need to get the transferer's connected line information copied
+		 * at this point because he is likely to hang up during the bridge with
+		 * newchan. This info will be used down below before bridging the 
+		 * transferee and newchan
+		 *
+		 * As a result, we need to be sure to free this data before returning
+		 * or overwriting it.
+		 */
+		ast_channel_lock(transferer);
+		ast_party_connected_line_copy(&connected_line, &transferer->connected);
+		ast_channel_unlock(transferer);
 		res = ast_bridge_call(transferer, newchan, &bconfig);
 		if (ast_check_hangup(newchan) || !ast_check_hangup(transferer)) {
 			ast_hangup(newchan);
@@ -1485,10 +1497,12 @@ static int builtin_atxfer(struct ast_channel *chan, struct ast_channel *peer, st
 				ast_log(LOG_WARNING, "Failed to play transfer sound!\n");
 			finishup(transferee);
 			transferer->_softhangup = 0;
+			ast_party_connected_line_free(&connected_line);
 			return AST_FEATURE_RETURN_SUCCESS;
 		}
 		if (check_compat(transferee, newchan)) {
 			finishup(transferee);
+			ast_party_connected_line_free(&connected_line);
 			return -1;
 		}
 		ast_indicate(transferee, AST_CONTROL_UNHOLD);
@@ -1499,11 +1513,13 @@ static int builtin_atxfer(struct ast_channel *chan, struct ast_channel *peer, st
 		 || ast_check_hangup(transferee)
 		 || ast_check_hangup(newchan)) {
 			ast_hangup(newchan);
+			ast_party_connected_line_free(&connected_line);
 			return -1;
 		}
 		xferchan = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, "", "", "", 0, "Transfered/%s", transferee->name);
 		if (!xferchan) {
 			ast_hangup(newchan);
+			ast_party_connected_line_free(&connected_line);
 			return -1;
 		}
 		/* Make formats okay */
@@ -1523,6 +1539,7 @@ static int builtin_atxfer(struct ast_channel *chan, struct ast_channel *peer, st
 		if (!(tobj = ast_calloc(1, sizeof(*tobj)))) {
 			ast_hangup(xferchan);
 			ast_hangup(newchan);
+			ast_party_connected_line_free(&connected_line);
 			return -1;
 		}
 
@@ -1557,6 +1574,18 @@ static int builtin_atxfer(struct ast_channel *chan, struct ast_channel *peer, st
 			tobj->bconfig.end_bridge_callback_data_fixup(&tobj->bconfig, tobj->peer, tobj->chan);
 		}
 
+		/* Due to a limitation regarding when callerID is set on a Local channel,
+		 * we use the transferer's connected line information here.
+		 */
+		connected_line.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
+		ast_channel_update_connected_line(xferchan, &connected_line);
+		ast_channel_lock(xferchan);
+		ast_connected_line_copy_from_caller(&connected_line, &xferchan->cid);
+		ast_channel_unlock(xferchan);
+		connected_line.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
+		ast_channel_update_connected_line(newchan, &connected_line);
+		ast_party_connected_line_free(&connected_line);
+
 		if (ast_stream_and_wait(newchan, xfersound, ""))
 			ast_log(LOG_WARNING, "Failed to play transfer sound!\n");
 		bridge_call_thread_launch(tobj);
@@ -1653,11 +1682,24 @@ static int builtin_atxfer(struct ast_channel *chan, struct ast_channel *peer, st
 		tobj->chan = newchan;
 		tobj->peer = xferchan;
 		tobj->bconfig = *config;
-
+		
 		if (tobj->bconfig.end_bridge_callback_data_fixup) {
 			tobj->bconfig.end_bridge_callback_data_fixup(&tobj->bconfig, tobj->peer, tobj->chan);
 		}
 
+		ast_channel_lock(newchan);
+		ast_connected_line_copy_from_caller(&connected_line, &newchan->cid);
+		ast_channel_unlock(newchan);
+		connected_line.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
+		ast_channel_update_connected_line(xferchan, &connected_line);
+		ast_channel_lock(xferchan);
+		ast_connected_line_copy_from_caller(&connected_line, &xferchan->cid);
+		ast_channel_unlock(xferchan);
+		connected_line.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
+		ast_channel_update_connected_line(newchan, &connected_line);
+
+		ast_party_connected_line_free(&connected_line);
+		
 		if (ast_stream_and_wait(newchan, xfersound, ""))
 			ast_log(LOG_WARNING, "Failed to play transfer sound!\n");
 		bridge_call_thread_launch(tobj);
@@ -2197,6 +2239,10 @@ static struct ast_channel *feature_request_and_dial(struct ast_channel *caller,
 	ast_channel_inherit_variables(caller, chan);	
 	pbx_builtin_setvar_helper(chan, "TRANSFERERNAME", caller->name);
 		
+	ast_channel_lock(chan);
+	ast_connected_line_copy_from_caller(&chan->connected, &caller->cid);
+	ast_channel_unlock(chan);
+	
 	if (ast_call(chan, data, timeout)) {
 		ast_log(LOG_NOTICE, "Unable to call channel %s/%s\n", type, (char *)data);
 		goto done;
@@ -2266,6 +2312,8 @@ static struct ast_channel *feature_request_and_dial(struct ast_channel *caller,
 					f = NULL;
 					ready=1;
 					break;
+				} else if (f->subclass == AST_CONTROL_CONNECTED_LINE) {
+					ast_indicate_data(caller, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
 				} else if (f->subclass != -1) {
 					ast_log(LOG_NOTICE, "Don't know what to do about control frame: %d\n", f->subclass);
 				}
@@ -4444,8 +4492,19 @@ int ast_pickup_call(struct ast_channel *chan)
 	struct ast_channel *cur = ast_channel_search_locked(find_channel_by_group, chan);
 
 	if (cur) {
+		struct ast_party_connected_line connected_caller;
+
 		int res = -1;
 		ast_debug(1, "Call pickup on chan '%s' by '%s'\n",cur->name, chan->name);
+
+		connected_caller = cur->connected;
+		connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
+		ast_channel_update_connected_line(chan, &connected_caller);
+
+		ast_party_connected_line_collect_caller(&connected_caller, &chan->cid);
+		connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
+		ast_channel_queue_connected_line_update(chan, &connected_caller);
+
 		res = ast_answer(chan);
 		if (res)
 			ast_log(LOG_WARNING, "Unable to answer '%s'\n", chan->name);