Add option to not do a call forward on 482 Loop Detected

Asterisk has always set up a forwarded call when receiving a 482 Loop Detected.
This prevents handling the call failure by just continuing on in the dialplan.
Since this would be a change in behavior, the new option to disable this
behavior is forwardloopdetected which defaults to 'yes'.

Review: https://reviewboard.asterisk.org/r/764/


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@274280 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 19 insertions, 10 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f2585a487..01742f8cf 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -812,11 +812,12 @@ struct sip_auth {
 #define SIP_PAGE2_UDPTL_DESTINATION     (1 << 28)       /*!< 28: Use source IP of RTP as destination if NAT is enabled */
 #define SIP_PAGE2_DIALOG_ESTABLISHED    (1 << 29)       /*!< 29: Has a dialog been established? */
 #define SIP_PAGE2_RPORT_PRESENT         (1 << 30)       /*!< 30: Was rport received in the Via header? */
+#define SIP_PAGE2_FORWARD_LOOP_DETECTED (1 << 31)       /*!< 31: Do call forward when receiving 482 Loop Detected */
 
 #define SIP_PAGE2_FLAGS_TO_COPY \
 	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
 	SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
-	SIP_PAGE2_UDPTL_DESTINATION)
+	SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_FORWARD_LOOP_DETECTED)
 
 /* SIP packet flags */
 #define SIP_PKT_DEBUG		(1 << 0)	/*!< Debug this packet */
@@ -11200,6 +11201,7 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, const struct m
 		ast_cli(fd, "  Send RPID    : %s\n", ast_test_flag(&peer->flags[0], SIP_SENDRPID) ? "Yes" : "No");
 		ast_cli(fd, "  Subscriptions: %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE) ? "Yes" : "No");
 		ast_cli(fd, "  Overlap dial : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP) ? "Yes" : "No");
+		ast_cli(fd, "  Forward Loop : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED) ? "Yes" : "No");
 
 		/* - is enumerated */
 		ast_cli(fd, "  DTMFmode     : %s\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
@@ -11503,7 +11505,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
 	ast_cli(fd, "  MOH Interpret:          %s\n", default_mohinterpret);
 	ast_cli(fd, "  MOH Suggest:            %s\n", default_mohsuggest);
 	ast_cli(fd, "  Voice Mail Extension:   %s\n", default_vmexten);
-
+	ast_cli(fd, "  Forward Detected Loops: %s\n", (ast_test_flag(&global_flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED) ? "Yes" : "No"));
 	
 	if (realtimepeers || realtimeusers) {
 		ast_cli(fd, "\nRealtime SIP Settings:\n");
@@ -13701,16 +13703,19 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
 					if (p->owner)
 						ast_queue_control(p->owner, AST_CONTROL_BUSY);
 					break;
-				case 482: /*
-					\note SIP is incapable of performing a hairpin call, which
-					is yet another failure of not having a layer 2 (again, YAY
-					 IETF for thinking ahead).  So we treat this as a call
-					 forward and hope we end up at the right place... */
-					if (option_debug)
-						ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n");
-					if (p->owner)
+				case 482: /* Loop Detected */
+					/*
+						\note Asterisk has historically tried to do a call forward when it
+						gets a 482, but that behavior isn't necessarily the best course of
+						action. Go ahead and do it anyway by default, but allow the option
+						to immediately pass to the next line in the dialplan. */
+					if (p->owner && ast_test_flag(&p->flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED)) {
+						if (option_debug) {
+							ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n");
+						}
 						ast_string_field_build(p->owner, call_forward,
 								       "Local/%s@%s", p->username, p->context);
+					}
 					/* Fall through */
 				case 480: /* Temporarily Unavailable */
 				case 404: /* Not Found */
@@ -17570,6 +17575,9 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
 	} else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
+	} else if (!strcasecmp(v->name, "forwardloopdetected")) {
+		ast_set_flag(&mask[1], SIP_PAGE2_FORWARD_LOOP_DETECTED);
+		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_FORWARD_LOOP_DETECTED);
 	} else
 		res = 0;
 
@@ -18386,6 +18394,7 @@ static int reload_config(enum channelreloadreason reason)
 	ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833);			/*!< Default DTMF setting: RFC2833 */
 	ast_set_flag(&global_flags[0], SIP_NAT_RFC3581);			/*!< NAT support if requested by device with rport */
 	ast_set_flag(&global_flags[0], SIP_CAN_REINVITE);			/*!< Allow re-invites */
+	ast_set_flag(&global_flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED); /*!< Set up call forward on 482 Loop Detected */
 
 	/* Debugging settings, always default to off */
 	dumphistory = FALSE;