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Merge Mike's announcement enhancements
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -418,6 +418,7 @@ static int dial_exec(struct ast_channel *chan, void *data)
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char *stack,*var;
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int play_to_caller=0,play_to_callee=0;
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int playargs=0;
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int digit = 0;
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if (!data) {
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ast_log(LOG_WARNING, "Dial requires an argument (technology1/number1&technology2/number2...|optional timeout|options)\n");
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@@ -837,11 +838,16 @@ static int dial_exec(struct ast_channel *chan, void *data)
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// Now Stream the File
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if (!res)
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res = ast_streamfile(peer,announcemsg,peer->language);
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if (!res)
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res = ast_waitstream(peer,"");
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if (!res) {
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digit = ast_waitstream(peer, AST_DIGIT_ANY);
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}
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// Ok, done. stop autoservice
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res = ast_autoservice_stop(chan);
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if (digit > 0 && !res)
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res = ast_senddigit(chan, digit);
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else
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res = digit;
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} else
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res = 0;
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