Detect potential forwarding loops based on count.

A potential problem that can arise is the following:

* Bob's phone is programmed to automatically forward to Carol.
* Carol's phone is programmed to automatically forward to Bob.
* Alice calls Bob.

If left unchecked, this results in an endless loops of call forwards
that would eventually result in some sort of fiery crash.

Asterisk's method of solving this issue was to track which interfaces
had been dialed. If a destination were dialed a second time, then
the attempt to call that destination would fail since a loop was
detected.

The problem with this method is that call forwarding has evolved. Some
SIP phones allow for a user to manually forward an incoming call to an
ad-hoc destination. This can mean that:

* There are legitimate use cases where a device may be dialed multiple
times, or
* There can be human error when forwarding calls.

This change removes the old method of detecting forwarding loops in
favor of keeping a count of the number of destinations a channel has
dialed on a particular branch of a call. If the number exceeds the
set number of max forwards, then the call fails. This approach has
the following advantages over the old:

* It is much simpler.
* It can detect loops involving local channels.
* It is user configurable.

The only disadvantage it has is that in the case where there is a
legitimate forwarding loop present, it takes longer to detect it.
However, the forwarding loop is still properly detected and the
call is cleaned up as it should be.

Address review feedback on gerrit.

* Correct "mfgium" to "Digium"
* Decrement max forwards by one in the case where allocation of the
  max forwards datastore is required.
* Remove irrelevant code change from pjsip_global_headers.c

ASTERISK-24958 #close

Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23

1 files changed, 3 insertions, 25 deletions

diff --git a/main/features.c b/main/features.c
index 971fb4a02..4acd8aab2 100644
--- a/main/features.c
+++ b/main/features.c
@@ -78,6 +78,7 @@ ASTERISK_REGISTER_FILE()
 #include "asterisk/stasis.h"
 #include "asterisk/stasis_channels.h"
 #include "asterisk/features_config.h"
+#include "asterisk/max_forwards.h"
 
 /*** DOCUMENTATION
 	<application name="Bridge" language="en_US">
@@ -420,22 +421,6 @@ static void add_features_datastores(struct ast_channel *caller, struct ast_chann
 	add_features_datastore(callee, &config->features_callee, &config->features_caller);
 }
 
-static void clear_dialed_interfaces(struct ast_channel *chan)
-{
-	struct ast_datastore *di_datastore;
-
-	ast_channel_lock(chan);
-	if ((di_datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL))) {
-		if (option_debug) {
-			ast_log(LOG_DEBUG, "Removing dialed interfaces datastore on %s since we're bridging\n", ast_channel_name(chan));
-		}
-		if (!ast_channel_datastore_remove(chan, di_datastore)) {
-			ast_datastore_free(di_datastore);
-		}
-	}
-	ast_channel_unlock(chan);
-}
-
 static void bridge_config_set_limits_warning_values(struct ast_bridge_config *config, struct ast_bridge_features_limits *limits)
 {
 	if (config->end_sound) {
@@ -572,20 +557,13 @@ static int pre_bridge_setup(struct ast_channel *chan, struct ast_channel *peer,
 	ast_channel_log("Pre-bridge PEER Channel info", peer);
 #endif
 
-	/*
-	 * If we are bridging a call, stop worrying about forwarding
-	 * loops.  We presume that if a call is being bridged, that the
-	 * humans in charge know what they're doing.  If they don't,
-	 * well, what can we do about that?
-	 */
-	clear_dialed_interfaces(chan);
-	clear_dialed_interfaces(peer);
-
 	res = 0;
 	ast_channel_lock(chan);
+	ast_max_forwards_reset(chan);
 	res |= ast_bridge_features_ds_append(chan, &config->features_caller);
 	ast_channel_unlock(chan);
 	ast_channel_lock(peer);
+	ast_max_forwards_reset(peer);
 	res |= ast_bridge_features_ds_append(peer, &config->features_callee);
 	ast_channel_unlock(peer);