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>
CERROR(call, NULL, "Linked call structure has no link to us.\n");
call->bridge_call->bridge_call = NULL;
}
+ if (call->trans)
+ ast_translator_free_path(call->trans);
+ if (call->dsp)
+ ast_dsp_free(call->dsp);
CDEBUG(call, NULL, "Call instance freed.\n");
free(call);
return;
{
union parameter newparam;
char string[1024], *p = string, *opt, *key;
- int gain, i, newmode = 0;
+ int gain, i;
if (!data[0])
return; // no opts
break;
}
CDEBUG(call, call->ast, "Option 'h' (HDLC).\n");
- if (!call->hdlc) {
+ if (!call->hdlc)
call->hdlc = 1;
- newmode = 1;
- }
break;
case 't':
if (opt[1] != '\0') {
break;
}
CDEBUG(call, call->ast, "Option 't' (no dsp).\n");
- if (!call->nodsp) {
+ if (!call->nodsp)
call->nodsp = 1;
- newmode = 1;
- }
break;
case 'e':
if (opt[1] == '\0') {
if (call->bchannel)
bchannel_pipeline(call->bchannel, call->pipeline);
break;
+ case 'f':
+ if (opt[1] == '\0') {
+ CERROR(call, call->ast, "Option 'f' (faxdetect) expects parameter.\n", opt);
+ break;
+ }
+ call->faxdetect=atoi(opt+1);
+ if (!call->dsp)
+ call->dsp=ast_dsp_new();
+ if (call->dsp) {
+ ast_dsp_set_features(call->dsp, DSP_FEATURE_DTMF_DETECT| DSP_FEATURE_FAX_DETECT);
+ if (!call->trans)
+ call->trans=ast_translator_build_path(AST_FORMAT_SLINEAR, (options.law=='a')?AST_FORMAT_ALAW:AST_FORMAT_ULAW);
+ }
+ CDEBUG(call, call->ast, "Option 'f' (faxdetect) with config '%s'.\n", call->faxdetect);
+ break;
case 'r':
if (opt[1] != '\0') {
CERROR(call, call->ast, "Option 'r' (re-buffer 160 bytes) expects no parameter.\n", opt);
}
CDEBUG(call, call->ast, "Option 'r' (re-buffer 160 bytes)");
call->rebuffer = 1;
+ call->framepos = 0;
break;
case 's':
if (opt[1] != '\0') {
/* re-open, if bchannel is created */
if (call->bchannel && call->bchannel->b_sock > -1) {
bchannel_destroy(call->bchannel);
- if (bchannel_create(call->bchannel, ((call->nodsp)?1:0) + ((call->hdlc)?2:0)))
+ if (bchannel_create(call->bchannel, ((call->nodsp || call->faxdetect > 0)?1:0) + ((call->hdlc)?2:0)))
bchannel_activate(call->bchannel, 1);
}
}
}
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);
}
}
bchannel_join(bchannel, call->bridge_id);
}
/* create only, if call exists, othewhise it bchannel is freed below... */
- if (bchannel_create(bchannel, ((call->nodsp)?1:0) + ((call->hdlc)?2:0)))
+ if (bchannel_create(bchannel, ((call->nodsp || call->faxdetect > 0)?1:0) + ((call->hdlc)?2:0)))
bchannel_activate(bchannel, 1);
}
/* acknowledge */
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");
/* 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))
return -1;
}
if (call->bchannel && f->samples)
- bchannel_transmit(call->bchannel, f->data, f->samples);
+ bchannel_transmit(call->bchannel, (unsigned char *)f->data, f->samples);
ast_mutex_unlock(&chan_lock);
return 0;
}
}
if (call->pipe[0] > -1) {
if (call->rebuffer && !call->hdlc) {
- len = read(call->pipe[0], call->read_buff, 160);
+ /* Make sure we have a complete 20ms (160byte) frame */
+ len=read(call->pipe[0],call->read_buff + call->framepos, 160 - call->framepos);
+ if (len > 0) {
+ call->framepos += len;
+ }
} else {
len = read(call->pipe[0], call->read_buff, sizeof(call->read_buff));
}
call->pipe[0] = -1;
ast_mutex_unlock(&chan_lock);
return NULL;
+ } else if (call->rebuffer && call->framepos < 160) {
+ /* Not a complete frame, so we send a null-frame */
+ ast_mutex_unlock(&chan_lock);
+ return &ast_null_frame;
}
}
call->read_fr.frametype = AST_FRAME_VOICE;
call->read_fr.subclass = ast->nativeformats;
- call->read_fr.datalen = len;
- call->read_fr.samples = len;
+ if (call->rebuffer) {
+ call->read_fr.datalen = call->framepos;
+ call->read_fr.samples = call->framepos;
+ call->framepos = 0;
+ } else {
+ call->read_fr.datalen = len;
+ call->read_fr.samples = len;
+ }
call->read_fr.delivery = ast_tv(0,0);
- call->read_fr.data = call->read_buff;
+ *((unsigned char **)&(call->read_fr.data)) = call->read_buff;
ast_mutex_unlock(&chan_lock);
return &call->read_fr;
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;
return AST_BRIDGE_COMPLETE;
}
- /* join, if both call instances uses dsp */
+ /* join, if both call instances uses dsp
+ ignore the case of fax detection here it may be benificial for ISDN fax machines or pass through.
+ */
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);
}
static struct ast_channel_tech lcr_tech = {
.type="LCR",
- .description="Channel driver for connecting to Linux-Call-Router",
- .requester=lcr_request,
- .send_digit_begin=lcr_digit_begin,
- .send_digit_end=lcr_digit_end,
- .call=lcr_call,
- .bridge=lcr_bridge,
- .hangup=lcr_hangup,
- .answer=lcr_answer,
- .read=lcr_read,
- .write=lcr_write,
- .indicate=lcr_indicate,
- .fixup=lcr_fixup,
- .send_text=lcr_send_text,
- .properties=0
+ .description = "Channel driver for connecting to Linux-Call-Router",
+ .capabilities = AST_FORMAT_ALAW,
+ .requester = lcr_request,
+ .send_digit_begin = lcr_digit_begin,
+ .send_digit_end = lcr_digit_end,
+ .call = lcr_call,
+ .bridge = lcr_bridge,
+ .hangup = lcr_hangup,
+ .answer = lcr_answer,
+ .read = lcr_read,
+ .write = lcr_write,
+ .indicate = lcr_indicate,
+ .fixup = lcr_fixup,
+ .send_text = lcr_send_text,
+ .properties = 0
};
" n - Don't detect dtmf tones on called channel.\n"
" h - Force data call (HDLC).\n"
" t - Disable mISDN_dsp features (required for fax application).\n"
+ " f - Adding fax detection. It it timeouts, mISDN_dsp is used.\n"
+ " Use time to detect for optarg.\n"
" 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"
+ " r - re-buffer packets (160 bytes). Required for some SIP-phones and fax applications.\n"
" vr - rxgain control\n"
" vt - txgain control\n"
" Volume changes at factor 2 ^ optarg.\n"
projects / lcr.git / blobdiff