From: Automerge script <automerge@asterisk.org>
Date: Thu, 4 Oct 2012 18:23:00 +0000 (+0000)
Subject: Merged revisions 374457 via svnmerge from
X-Git-Tag: 10.10.0-digiumphones-rc1~8
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=afa876c30302b7f74c666e754b3b80dcc78a3ef5;p=thirdparty%2Fasterisk.git

Merged revisions 374457 via svnmerge from
file:///srv/subversion/repos/asterisk/branches/10

................
  r374457 | file | 2012-10-04 12:44:38 -0500 (Thu, 04 Oct 2012) | 17 lines

  Fix a regression from direct media ACLs where the directrtpsetup option no longer works.

  A check was added for direct media ACLs that immediately forbid remote bridging if there
  was no bridged channel. This caused directrtpsetup to no longer function as it needs this
  information before bridging actually occurs.

  Logic has now been adjusted so if there is no bridged channel a remote bridge will still
  be attempted.

  (closes issue ASTERISK-20511)
  Reported by: kristoff

  Review: https://reviewboard.asterisk.org/r/2146/
  ........

  Merged revisions 374456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index def30b5e87..725314702f 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -30489,22 +30489,22 @@ static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struc
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
 
-	if (!(opp_chan = ast_bridged_channel(chan))) {
-		return AST_RTP_GLUE_RESULT_FORBID;
-	} else if (((opp_chan->tech != &sip_tech) && (opp_chan->tech != &sip_tech_info)) ||
-		   (!(opp = opp_chan->tech_pvt))) {
+	if ((opp_chan = ast_bridged_channel(chan)) && (((opp_chan->tech != &sip_tech) && (opp_chan->tech != &sip_tech_info)) ||
+						       (!(opp = opp_chan->tech_pvt)))) {
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
 
 	sip_pvt_lock(p);
-	while (sip_pvt_trylock(opp)) {
+	while (opp && sip_pvt_trylock(opp)) {
 		sip_pvt_unlock(p);
 		usleep(1);
 		sip_pvt_lock(p);
 	}
 
 	if (!(p->rtp)) {
-		sip_pvt_unlock(opp);
+		if (opp) {
+			sip_pvt_unlock(opp);
+		}
 		sip_pvt_unlock(p);
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
@@ -30514,7 +30514,7 @@ static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struc
 
 	if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
 		res = AST_RTP_GLUE_RESULT_REMOTE;
-		if (!apply_directmedia_ha(p, opp, "audio")) {
+		if (opp && !apply_directmedia_ha(p, opp, "audio")) {
 			res = AST_RTP_GLUE_RESULT_FORBID;
 		}
 	} else if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) {
@@ -30523,7 +30523,9 @@ static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struc
 		res = AST_RTP_GLUE_RESULT_FORBID;
 	}
 
-	sip_pvt_unlock(opp);
+	if (opp) {
+		sip_pvt_unlock(opp);
+	}
 
 	if (p->srtp) {
 		res = AST_RTP_GLUE_RESULT_FORBID;
@@ -30545,22 +30547,22 @@ static enum ast_rtp_glue_result sip_get_vrtp_peer(struct ast_channel *chan, stru
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
 
-	if (!(opp_chan = ast_bridged_channel(chan))) {
-		return AST_RTP_GLUE_RESULT_FORBID;
-	} else if (((opp_chan->tech != &sip_tech) && (opp_chan->tech != &sip_tech_info)) ||
-		   (!(opp = opp_chan->tech_pvt))) {
+	if ((opp_chan = ast_bridged_channel(chan)) && (((opp_chan->tech != &sip_tech) && (opp_chan->tech != &sip_tech_info)) ||
+						       (!(opp = opp_chan->tech_pvt)))) {
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
 
 	sip_pvt_lock(p);
-	while (sip_pvt_trylock(opp)) {
+	while (opp && sip_pvt_trylock(opp)) {
 		sip_pvt_unlock(p);
 		usleep(1);
 		sip_pvt_lock(p);
 	}
 
 	if (!(p->vrtp)) {
-		sip_pvt_unlock(opp);
+		if (opp) {
+			sip_pvt_unlock(opp);
+		}
 		sip_pvt_unlock(p);
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
@@ -30570,12 +30572,14 @@ static enum ast_rtp_glue_result sip_get_vrtp_peer(struct ast_channel *chan, stru
 
 	if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
 		res = AST_RTP_GLUE_RESULT_REMOTE;
-		if (!apply_directmedia_ha(p, opp, "video")) {
+		if (opp && !apply_directmedia_ha(p, opp, "video")) {
 			res = AST_RTP_GLUE_RESULT_FORBID;
 		}
 	}
 
-	sip_pvt_unlock(opp);
+	if (opp) {
+		sip_pvt_unlock(opp);
+	}
 	sip_pvt_unlock(p);
 
 	return res;
@@ -30592,22 +30596,22 @@ static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, stru
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
 
-	if (!(opp_chan = ast_bridged_channel(chan))) {
-		return AST_RTP_GLUE_RESULT_FORBID;
-	} else if (((opp_chan->tech != &sip_tech) && (opp_chan->tech != &sip_tech_info)) ||
-		   (!(opp = opp_chan->tech_pvt))) {
+	if ((opp_chan = ast_bridged_channel(chan)) && (((opp_chan->tech != &sip_tech) && (opp_chan->tech != &sip_tech_info)) ||
+						       (!(opp = opp_chan->tech_pvt)))) {
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
 
 	sip_pvt_lock(p);
-	while (sip_pvt_trylock(opp)) {
+	while (opp && sip_pvt_trylock(opp)) {
 		sip_pvt_unlock(p);
 		usleep(1);
 		sip_pvt_lock(p);
 	}
 
 	if (!(p->trtp)) {
-		sip_pvt_unlock(opp);
+		if (opp) {
+			sip_pvt_unlock(opp);
+		}
 		sip_pvt_unlock(p);
 		return AST_RTP_GLUE_RESULT_FORBID;
 	}
@@ -30617,12 +30621,14 @@ static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, stru
 
 	if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
 		res = AST_RTP_GLUE_RESULT_REMOTE;
-		if (!apply_directmedia_ha(p, opp, "text")) {
+		if (opp && !apply_directmedia_ha(p, opp, "text")) {
 			res = AST_RTP_GLUE_RESULT_FORBID;
 		}
 	}
 
-	sip_pvt_unlock(opp);
+	if (opp) {
+		sip_pvt_unlock(opp);
+	}
 	sip_pvt_unlock(p);
 
 	return res;