Merged revisions 53052 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2 lines

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.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53053 65c4cc65-6c06-0410-ace0-fbb531ad65f3

main/rtp.c

@@ -2832,7 +2832,7 @@ 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 */
@@ -2951,6 +2951,19 @@ 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 {