Call is initiated by LCR:
If a call is received from LCR, a MESSAGE_NEWREF is received first.
+The ref_was_assigned ist set to 1.
A new chan_call instance is created. The call reference (ref) is given by
-MESSAGE_NEWREF. The state is CHAN_LCR_STATE_IN_PREPARE.
+the received MESSAGE_NEWREF. The state is CHAN_LCR_STATE_IN_PREPARE.
After receiving MESSAGE_SETUP from LCR, the ast_channel instance is created
using ast_channel_alloc(1). The setup information is given to asterisk.
The new Asterisk instance pointer (ast) is stored to chan_call structure.
If a call is requested from Asterisk, a new chan_call instance is created.
The new Asterisk instance pointer (ast) is stored to chan_call structure.
The current call ref is set to 0, the state is CHAN_LCR_STATE_OUT_PREPARE.
-If the call is received (lcr_call) A MESSASGE_NEWREF is sent to LCR requesting
+If the call is received (lcr_call) A MESSAGE_NEWREF is sent to LCR requesting
a new call reference (ref).
+The ref_was_assigned ist set to 1.
Further dialing information is queued.
After the new callref is received by special MESSAGE_NEWREF reply, new ref
is stored in the chan_call structure.
A MESSAGE_RELEASE is received with the call reference (ref) to be released.
The current ref is set to 0, to indicate released reference.
+The ref_was_assigned==1 shows that there is no other ref to be assigned.
The state changes to CHAN_LCR_STATE_RELEASE.
ast_queue_hangup() is called, if asterisk instance (ast) exists, if not,
the chan_call instance is destroyed.
a MESSAGE_RELEASE is sent to LCR and the chan_call instance is destroyed.
If the ref is 0 and the state is not CHAN_LCR_STATE_RELEASE, the new state is
set to CHAN_LCR_STATE_RELEASE.
+The ref_was_assigned==0 shows that a ref is still requested.
Later, if the MESSAGE_NEWREF reply is received, a MESSAGE_RELEASE is sent to
LCR and the chan_call instance is destroyed.
If the ref is 0 and the state is CHAN_LCR_STATE_RELEASE, see the proceedure
#include <semaphore.h>
+#define HAVE_ATTRIBUTE_always_inline 1
+#define HAVE_ARPA_INET_H 1
+#define HAVE_TIMERSUB 1
+
+#include <asterisk/compiler.h>
+#include <asterisk/buildopts.h>
#include <asterisk/module.h>
#include <asterisk/channel.h>
#include <asterisk/config.h>
CDEBUG(call, call->ast, "Option 'r' (re-buffer 160 bytes)");
call->rebuffer = 1;
break;
-#if 0
case 's':
if (opt[1] != '\0') {
CERROR(call, call->ast, "Option 's' (inband DTMF) expects no parameter.\n", opt);
}
CDEBUG(call, call->ast, "Option 's' (inband DTMF).\n");
call->inband_dtmf = 1;
-todo
break;
-#endif
case 'v':
if (opt[1] != 'r' && opt[1] != 't') {
CERROR(call, call->ast, "Option 'v' (volume) expects parameter.\n", opt);
}
switch(ast->cid.cid_pres & AST_PRES_RESTRICTION)
{
- case AST_PRES_ALLOWED:
- newparam.setup.callerinfo.present = INFO_PRESENT_ALLOWED;
- break;
case AST_PRES_RESTRICTED:
newparam.setup.callerinfo.present = INFO_PRESENT_RESTRICTED;
break;
case AST_PRES_UNAVAILABLE:
newparam.setup.callerinfo.present = INFO_PRESENT_NOTAVAIL;
break;
+ case AST_PRES_ALLOWED:
default:
- newparam.setup.callerinfo.present = INFO_PRESENT_NULL;
+ newparam.setup.callerinfo.present = INFO_PRESENT_ALLOWED;
}
switch(ast->cid.cid_ton)
{
/* check bridge */
if (!call) return;
if (!call->bridge_call) return;
- CDEBUG(call, NULL, "Sending message due briding.\n");
+ CDEBUG(call, NULL, "Sending message due bridging.\n");
send_message(message_type, call->bridge_call->ref, param);
}
{
int cause, ret;
union parameter newparam;
+ char *exten = ast->exten;
+ if (!*exten)
+ exten = "s";
- CDEBUG(call, ast, "Try to start pbx. (exten=%s context=%s complete=%s)\n", ast->exten, ast->context, complete?"yes":"no");
+ CDEBUG(call, ast, "Try to start pbx. (exten=%s context=%s complete=%s)\n", exten, ast->context, complete?"yes":"no");
if (complete)
{
/* if not match */
- if (!ast_canmatch_extension(ast, ast->context, ast->exten, 1, call->oad))
+ if (!ast_canmatch_extension(ast, ast->context, exten, 1, call->oad))
{
- CDEBUG(call, ast, "Got 'sending complete', but extension '%s' will not match at context '%s' - releasing.\n", ast->exten, ast->context);
+ CDEBUG(call, ast, "Got 'sending complete', but extension '%s' will not match at context '%s' - releasing.\n", exten, ast->context);
cause = 1;
goto release;
}
- if (!ast_exists_extension(ast, ast->context, ast->exten, 1, call->oad))
+ if (!ast_exists_extension(ast, ast->context, exten, 1, call->oad))
{
- CDEBUG(call, ast, "Got 'sending complete', but extension '%s' would match at context '%s', if more digits would be dialed - releasing.\n", ast->exten, ast->context);
+ CDEBUG(call, ast, "Got 'sending complete', but extension '%s' would match at context '%s', if more digits would be dialed - releasing.\n", exten, ast->context);
cause = 28;
goto release;
}
goto start;
}
- if (ast_canmatch_extension(ast, ast->context, ast->exten, 1, call->oad))
+ if (ast_canmatch_extension(ast, ast->context, exten, 1, call->oad))
{
/* send setup acknowledge to lcr */
if (call->state != CHAN_LCR_STATE_IN_DIALING) {
call->state = CHAN_LCR_STATE_IN_DIALING;
/* if match, start pbx */
- if (ast_exists_extension(ast, ast->context, ast->exten, 1, call->oad)) {
+ if (ast_exists_extension(ast, ast->context, exten, 1, call->oad)) {
CDEBUG(call, ast, "Extensions matches.\n");
goto start;
}
return;
}
+ if (!*ast->exten) {
+ /* if can match */
+ CDEBUG(call, ast, "There is no 's' extension (and we tried to match it implicitly). Extensions may match, if more digits are dialed.\n");
+ return;
+ }
+
/* if not match */
cause = 1;
release:
goto release;
}
call->pbx_started = 1;
-// if (call->state == CHAN_LCR_STATE_IN_DIALING)
- ast_setstate(ast, AST_STATE_RINGING);
-// else
-// ast_setstate(ast, AST_STATE_RINGING);
-// return;
+ ast_setstate(ast, AST_STATE_RING);
}
/*
/* use bridge to forware message not supported by asterisk */
if (call->state == CHAN_LCR_STATE_CONNECT) {
- CDEBUG(call, call->ast, "Call is connected, briding.\n");
+ CDEBUG(call, call->ast, "Call is connected, bridging.\n");
bridge_message_if_bridged(call, message_type, param);
}
}
case 'R':
CDEBUG(call, ast, "Sending queued RINGING to Asterisk.\n");
ast_queue_control(ast, AST_CONTROL_RINGING);
+ ast_setstate(ast, AST_STATE_RINGING);
break;
case 'A':
CDEBUG(call, ast, "Sending queued ANSWER to Asterisk.\n");
return 0;
}
+static void send_digit_to_chan(struct ast_channel * ast, char digit )
+{
+ static const char* dtmf_tones[] = {
+ "!941+1336/100,!0/100", /* 0 */
+ "!697+1209/100,!0/100", /* 1 */
+ "!697+1336/100,!0/100", /* 2 */
+ "!697+1477/100,!0/100", /* 3 */
+ "!770+1209/100,!0/100", /* 4 */
+ "!770+1336/100,!0/100", /* 5 */
+ "!770+1477/100,!0/100", /* 6 */
+ "!852+1209/100,!0/100", /* 7 */
+ "!852+1336/100,!0/100", /* 8 */
+ "!852+1477/100,!0/100", /* 9 */
+ "!697+1633/100,!0/100", /* A */
+ "!770+1633/100,!0/100", /* B */
+ "!852+1633/100,!0/100", /* C */
+ "!941+1633/100,!0/100", /* D */
+ "!941+1209/100,!0/100", /* * */
+ "!941+1477/100,!0/100" }; /* # */
+
+ if (digit >= '0' && digit <='9')
+ ast_playtones_start(ast,0,dtmf_tones[digit-'0'], 0);
+ else if (digit >= 'A' && digit <= 'D')
+ ast_playtones_start(ast,0,dtmf_tones[digit-'A'+10], 0);
+ else if (digit == '*')
+ ast_playtones_start(ast,0,dtmf_tones[14], 0);
+ else if (digit == '#')
+ ast_playtones_start(ast,0,dtmf_tones[15], 0);
+ else {
+ /* not handled */
+ ast_log(LOG_DEBUG, "Unable to handle DTMF tone "
+ "'%c' for '%s'\n", digit, ast->name);
+ }
+}
+
+
static int lcr_digit_begin(struct ast_channel *ast, char digit)
{
struct chan_call *call;
static int lcr_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
{
- printf("DIGIT END %c\n", digit);
+ int inband_dtmf = 0;
+ struct chan_call *call;
+
+ ast_mutex_lock(&chan_lock);
+
+ call = ast->tech_pvt;
+
+ if (!call) {
+ CERROR(NULL, ast,
+ "Received digit from Asterisk, "
+ "but no call instance exists.\n");
+ ast_mutex_unlock(&chan_lock);
+ return -1;
+ }
+
+ CDEBUG(call, ast, "DIGIT END '%c' from Asterisk.\n", digit);
+
+ if (call->state == CHAN_LCR_STATE_CONNECT && call->inband_dtmf) {
+ inband_dtmf = 1;
+ }
+
+ ast_mutex_unlock(&chan_lock);
+
+ if (inband_dtmf) {
+ CDEBUG(call, ast, "-> sending '%c' inband.\n", digit);
+ send_digit_to_chan(ast, digit);
+ }
+
return (0);
}
/* during prepare, we change to release state */
CDEBUG(call, ast, "We must wait until we received our ref, until we can free call instance.\n");
call->state = CHAN_LCR_STATE_RELEASE;
+ call->ast = NULL;
}
}
if (!pthread_equal(tid, chan_tid))
break;
#ifdef AST_CONTROL_SRCUPDATE
case AST_CONTROL_SRCUPDATE:
- CDEBUG(call, ast, "Received indicate AST_CONTROL_SRCUPDATE from Asterisk.\n");
- res = -1;
+ CDEBUG(call, ast, "Received AST_CONTROL_SRCUPDATE from Asterisk.\n");
break;
#endif
default:
struct ast_frame *f;
int bridge_id;
- CDEBUG(NULL, NULL, "Received briding request from Asterisk.\n");
+ CDEBUG(NULL, NULL, "Received bridging request from Asterisk.\n");
carr[0] = ast1;
carr[1] = ast2;
/* join, if both call instances uses dsp */
if (!call1->nodsp && !call2->nodsp) {
- CDEBUG(NULL, NULL, "Both calls use DSP, briding via DSP.\n");
+ CDEBUG(NULL, NULL, "Both calls use DSP, bridging via DSP.\n");
/* get bridge id and join */
bridge_id = new_bridge_id();
bchannel_join(call2->bchannel, bridge_id);
} else
if (call1->nodsp && call2->nodsp)
- CDEBUG(NULL, NULL, "Both calls use no DSP, briding in channel driver.\n");
+ CDEBUG(NULL, NULL, "Both calls use no DSP, bridging in channel driver.\n");
else
- CDEBUG(NULL, NULL, "One call uses no DSP, briding in channel driver.\n");
+ CDEBUG(NULL, NULL, "One call uses no DSP, bridging in channel driver.\n");
call1->bridge_call = call2;
call2->bridge_call = call1;
}
- CDEBUG(NULL, NULL, "Releasing bride.\n");
+ CDEBUG(NULL, NULL, "Releasing bridge.\n");
/* split channels */
ast_mutex_lock(&chan_lock);
" c - Make crypted outgoing call, optarg is keyindex.\n"
" e - Perform echo cancelation on this channel.\n"
" Takes mISDN pipeline option as optarg.\n"
-// " s - Send Non Inband DTMF as inband.\n"
+ " s - Send Non Inband DTMF as inband.\n"
" vr - rxgain control\n"
" vt - txgain control\n"
" Volume changes at factor 2 ^ optarg.\n"
projects / lcr.git / blobdiff