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Add SIP/RTP video support, video enable app_echo, start on RTCP
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -55,6 +55,9 @@ static int echo_exec(struct ast_channel *chan, void *data)
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if (f->frametype == AST_FRAME_VOICE) {
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if (ast_write(chan, f))
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break;
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} else if (f->frametype == AST_FRAME_VIDEO) {
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if (ast_write(chan, f))
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break;
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} else if (f->frametype == AST_FRAME_DTMF) {
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if (f->subclass == '#') {
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res = 0;
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