struct chan_call *next; /* link to next call instance */
int state; /* current call state CHAN_LCR_STATE */
unsigned int ref; /* callref for this channel */
+ int ref_was_assigned;
void *ast; /* current asterisk channel */
int pbx_started;
/* indicates if pbx que is available */
/* audio is available */
int cause, location;
/* store cause from lcr */
- unsigned char dialque[64];
+ char dialque[64];
/* queue dialing prior setup ack */
char oad[64];/* caller id in number format */
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 */
unsigned char bf_key[56];
int bf_len; /* blowfish crypt key */
- int transparent, 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 */
};