X-Git-Url: http://git.eversberg.eu/gitweb.cgi?p=lcr.git;a=blobdiff_plain;f=chan_lcr.h;h=5905ea1efabcc3fb58944a36fcb785cbc63f7399;hp=84932a271b8690c3e7dad06b3ef2d3f0dcf3aba0;hb=3ac6881c22bce18091f19b06958ac66016bf9a32;hpb=b519aa7e746912ab113b8332484418d0b2f00bcd

diff --git a/chan_lcr.h b/chan_lcr.h
index 84932a2..5905ea1 100644
--- a/chan_lcr.h
+++ b/chan_lcr.h
@@ -50,24 +50,33 @@ struct chan_call {
 	char                    cid_rdnis[64]; /* cached cid for setup */
 	char			display[128];
 					/* display for setup */
-	int			dtmf;
-					/* shall dtmf be enabled */
-	int			no_dtmf;
-					/* dtmf disabled by option */
+	int			dsp_dtmf;
+					/* decode dtmf by dsp */
+	int                     inband_dtmf; /* generate dtmf tones, if
+					      requested by asterisk */
         int                     rebuffer; /* send only 160 bytes frames
 					     to asterisk */
+
+	int			framepos; /* send only 160 bytes frames to asterisk */
+	
         int                     on_hold; /* track hold management, since
 					    sip phones sometimes screw it up */
 	char			pipeline[256];
 					/* echo cancel pipeline by option */
 	int			tx_gain, rx_gain;
 					/* gain by option */
+	int			keypad;
+					/* use keypad to dial number */
 	unsigned char		bf_key[56];
 	int			bf_len;	/* blowfish crypt key */
-	int			nodsp, hdlc;
+	struct ast_dsp		*dsp; /* ast dsp processor for fax/tone detection */
+	struct ast_trans_pvt 	*trans; /* Codec translation path as fax/tone detection requires slin */
+	int			nodsp, hdlc, faxdetect;
 					/* flags for bchannel mode */
 	char			queue_string[64];
 					/* queue for asterisk */
+	int			has_pattern;
+					/* pattern are available, PROGRESS has been indicated */
 		
 };