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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -82,8 +82,12 @@ struct ast_channel_tech {
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const char * const type;
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const char * const description;
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/*! Bitmap of formats this channel can handle */
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int capabilities;
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/*! Technology Properties */
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int properties;
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struct ast_channel *(* const requester)(const char *type, int format, void *data, int *cause);
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int (* const devicestate)(void *data);
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@@ -311,6 +315,10 @@ struct ast_channel {
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};
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/* Channel tech properties: */
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/* Channels have this property if they can accept input with jitter; i.e. most VoIP channels */
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#define AST_CHAN_TP_WANTSJITTER (1 << 0)
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#define AST_FLAG_DIGITAL (1 << 0) /* if the call is a digital ISDN call */
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#define AST_FLAG_DEFER_DTMF (1 << 1) /* if dtmf should be deferred */
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#define AST_FLAG_WRITE_INT (1 << 2) /* if write should be interrupt generator */
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@@ -21,6 +21,7 @@ extern "C" {
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#endif
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#include <asterisk/frame.h>
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#include <asterisk/plc.h>
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/* Declared by individual translators */
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struct ast_translator_pvt;
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@@ -103,7 +104,6 @@ extern void ast_translator_free_path(struct ast_trans_pvt *tr);
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*/
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extern struct ast_frame *ast_translate(struct ast_trans_pvt *tr, struct ast_frame *f, int consume);
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#if defined(__cplusplus) || defined(c_plusplus)
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}
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#endif
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