When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party.

git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53052 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 14 insertions, 1 deletions

diff --git a/main/rtp.c b/main/rtp.c
index fdf4ed68f..01b9e756d 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -2746,7 +2746,7 @@ static enum ast_bridge_result bridge_native_loop(struct ast_channel *c0, struct
 	int oldcodec0 = codec0, oldcodec1 = codec1;
 	struct sockaddr_in ac1 = {0,}, vac1 = {0,}, ac0 = {0,}, vac0 = {0,};
 	struct sockaddr_in t1 = {0,}, vt1 = {0,}, t0 = {0,}, vt0 = {0,};
-
+	
 	/* Set it up so audio goes directly between the two endpoints */
 
 	/* Test the first channel */
@@ -2864,6 +2864,19 @@ static enum ast_bridge_result bridge_native_loop(struct ast_channel *c0, struct
 			if ((fr->subclass == AST_CONTROL_HOLD) ||
 			    (fr->subclass == AST_CONTROL_UNHOLD) ||
 			    (fr->subclass == AST_CONTROL_VIDUPDATE)) {
+				if (fr->subclass == AST_CONTROL_HOLD) {
+					/* If we someone went on hold we want the other side to reinvite back to us */
+					if (who == c0)
+						pr1->set_rtp_peer(c1, NULL, NULL, 0, 0);
+					else
+						pr0->set_rtp_peer(c0, NULL, NULL, 0, 0);
+				} else if (fr->subclass == AST_CONTROL_UNHOLD) {
+					/* If they went off hold they should go back to being direct */
+					if (who == c0)
+						pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE));
+					else
+						pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE));
+				}
 				ast_indicate_data(other, fr->subclass, fr->data, fr->datalen);
 				ast_frfree(fr);
 			} else {