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authormarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2003-06-28 16:40:02 +0000
committermarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2003-06-28 16:40:02 +0000
commit63170c4333b792805df5170e43d516b479a75421 (patch)
tree350126d47aecc8eeeaca9794943d8d2640d37670 /apps/app_echo.c
parentefde31a9908909b35ee154afe257030a56a6c520 (diff)
Add SIP/RTP video support, video enable app_echo, start on RTCP
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1128 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'apps/app_echo.c')
-rwxr-xr-xapps/app_echo.c3
1 files changed, 3 insertions, 0 deletions
diff --git a/apps/app_echo.c b/apps/app_echo.c
index 7690209ac..27ba0d86f 100755
--- a/apps/app_echo.c
+++ b/apps/app_echo.c
@@ -55,6 +55,9 @@ static int echo_exec(struct ast_channel *chan, void *data)
if (f->frametype == AST_FRAME_VOICE) {
if (ast_write(chan, f))
break;
+ } else if (f->frametype == AST_FRAME_VIDEO) {
+ if (ast_write(chan, f))
+ break;
} else if (f->frametype == AST_FRAME_DTMF) {
if (f->subclass == '#') {
res = 0;