diff options
author | Matthew Nicholson <mnicholson@digium.com> | 2011-06-29 15:36:20 +0000 |
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committer | Matthew Nicholson <mnicholson@digium.com> | 2011-06-29 15:36:20 +0000 |
commit | 6c7d437287a662c8009ac8eb18b39aeb31f02060 (patch) | |
tree | afb00c4a2607562c313d485590e3870f33a79701 | |
parent | f99a06d030b95b45fe8209920613811b13830b79 (diff) |
Merged revisions 325537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) | 2 lines
don't do native/remote bridging if a framehook is active on the channel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rw-r--r-- | apps/app_dial.c | 4 | ||||
-rw-r--r-- | main/rtp_engine.c | 7 |
2 files changed, 8 insertions, 3 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c index 25172c743..d3b8009de 100644 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -65,6 +65,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/aoc.h" #include "asterisk/ccss.h" #include "asterisk/indications.h" +#include "asterisk/framehook.h" /*** DOCUMENTATION <application name="Dial" language="en_US"> @@ -631,7 +632,8 @@ END_OPTIONS ); OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \ OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | \ OPT_CALLER_PARK | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB) && \ - !chan->audiohooks && !peer->audiohooks) + !chan->audiohooks && !peer->audiohooks && \ + ast_framehook_list_is_empty(chan->framehooks) && ast_framehook_list_is_empty(peer->framehooks)) /* * The list of active channels diff --git a/main/rtp_engine.c b/main/rtp_engine.c index a1c460578..a3d9c4d8e 100644 --- a/main/rtp_engine.c +++ b/main/rtp_engine.c @@ -40,6 +40,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/translate.h" #include "asterisk/netsock2.h" #include "asterisk/_private.h" +#include "asterisk/framehook.h" struct ast_srtp_res *res_srtp = NULL; struct ast_srtp_policy_res *res_srtp_policy = NULL; @@ -853,7 +854,8 @@ static enum ast_bridge_result local_bridge_loop(struct ast_channel *c0, struct a if ((c0->tech_pvt != pvt0) || (c1->tech_pvt != pvt1) || (c0->masq || c0->masqr || c1->masq || c1->masqr) || - (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) { + (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks) || + (!ast_framehook_list_is_empty(c0->framehooks) || !ast_framehook_list_is_empty(c1->framehooks))) { ast_debug(1, "rtp-engine-local-bridge: Oooh, something is weird, backing out\n"); /* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */ if ((c0->masq || c0->masqr) && (fr = ast_read(c0))) { @@ -1046,7 +1048,8 @@ static enum ast_bridge_result remote_bridge_loop(struct ast_channel *c0, if ((c0->tech_pvt != pvt0) || (c1->tech_pvt != pvt1) || (c0->masq || c0->masqr || c1->masq || c1->masqr) || - (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks)) { + (c0->monitor || c0->audiohooks || c1->monitor || c1->audiohooks) || + (!ast_framehook_list_is_empty(c0->framehooks) || !ast_framehook_list_is_empty(c1->framehooks))) { ast_debug(1, "Oooh, something is weird, backing out\n"); res = AST_BRIDGE_RETRY; break; |