X-Git-Url: http://git.eversberg.eu/gitweb.cgi?p=lcr.git;a=blobdiff_plain;f=chan_lcr.h;h=5ddb4eb12a74a4fcaad60aa8ab012f2d0e3c5a6d;hp=0ed661e5021698cb6509249c3fe38b2e53bf8e70;hb=a425aedc1ee2c0bba4ba20904943afb21bd6e2e5;hpb=f51147e028fc6eb0db5cec7b6dbd685860bda5bd diff --git a/chan_lcr.h b/chan_lcr.h index 0ed661e..5ddb4eb 100644 --- a/chan_lcr.h +++ b/chan_lcr.h @@ -15,6 +15,7 @@ struct chan_call { 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 */ @@ -24,7 +25,7 @@ struct chan_call { /* 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 */ @@ -53,18 +54,29 @@ struct chan_call { /* shall dtmf be enabled */ int no_dtmf; /* dtmf disabled by option */ + 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 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 */ };