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Add SIP video fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -493,6 +493,11 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
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if (!ast_test_flag(outgoing, DIAL_RINGBACKONLY))
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ast_indicate(in, AST_CONTROL_PROGRESS);
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break;
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case AST_CONTROL_VIDUPDATE:
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if (option_verbose > 2)
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ast_verbose ( VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", o->chan->name,in->name);
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ast_indicate(in, AST_CONTROL_VIDUPDATE);
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break;
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case AST_CONTROL_PROCEEDING:
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if (option_verbose > 2)
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ast_verbose ( VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", o->chan->name,in->name);
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@@ -600,6 +605,11 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
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if (ast_write(outgoing->chan, f))
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ast_log(LOG_WARNING, "Unable to forward voice\n");
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}
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if (single && (f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_VIDUPDATE)) {
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if (option_verbose > 2)
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ast_verbose ( VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", in->name,outgoing->chan->name);
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ast_indicate(outgoing->chan, AST_CONTROL_VIDUPDATE);
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}
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ast_frfree(f);
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}
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if (!*to && (option_verbose > 2))
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