[Asterisk-cvs] asterisk/res res_features.c,1.57,1.58
kpfleming at lists.digium.com
kpfleming at lists.digium.com
Thu Jun 23 18:11:28 CDT 2005
Update of /usr/cvsroot/asterisk/res
In directory mongoose.digium.com:/tmp/cvs-serv23166/res
Modified Files:
res_features.c
Log Message:
support cancellation of attended transfers using the defined disconnect code (bug #3729 with minor mods)
Index: res_features.c
===================================================================
RCS file: /usr/cvsroot/asterisk/res/res_features.c,v
retrieving revision 1.57
retrieving revision 1.58
diff -u -d -r1.57 -r1.58
--- res_features.c 6 Jun 2005 22:12:19 -0000 1.57
+++ res_features.c 23 Jun 2005 22:12:01 -0000 1.58
@@ -32,6 +32,7 @@
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
+#include "asterisk/causes.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/app.h"
@@ -57,6 +58,8 @@
#define DEFAULT_TRANSFER_DIGIT_TIMEOUT 3000
#define DEFAULT_FEATURE_DIGIT_TIMEOUT 500
+#define AST_MAX_WATCHERS 256
+
static char *parkedcall = "ParkedCall";
/* No more than 45 seconds parked before you do something with them */
@@ -196,6 +199,9 @@
}
}
+static struct ast_channel *ast_feature_request_and_dial(struct ast_channel *caller, const char *type, int format, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name);
+
+
static void *ast_bridge_call_thread(void *data)
{
struct ast_bridge_thread_obj *tobj = data;
@@ -697,7 +703,9 @@
cid_name = transferer->cid.cid_name;
if (ast_exists_extension(transferer, transferer_real_context,xferto, 1, cid_num)) {
snprintf(dialstr, sizeof(dialstr), "%s@%s/n", xferto, transferer_real_context);
- if ((newchan = ast_request_and_dial("Local", ast_best_codec(transferer->nativeformats), dialstr,30000, &outstate, cid_num, cid_name))) {
+ newchan = ast_feature_request_and_dial(transferer, "Local", ast_best_codec(transferer->nativeformats), dialstr, 15000, &outstate, cid_num, cid_name);
+ ast_indicate(transferer, -1);
+ if(newchan){
res = ast_channel_make_compatible(transferer, newchan);
if (res < 0) {
ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", transferer->name, newchan->name);
@@ -708,7 +716,7 @@
ast_set_flag(&(bconfig.features_caller), AST_FEATURE_DISCONNECT);
ast_set_flag(&(bconfig.features_callee), AST_FEATURE_DISCONNECT);
res = ast_bridge_call(transferer,newchan,&bconfig);
- if (newchan->_softhangup || newchan->_state != AST_STATE_UP) {
+ if (newchan->_softhangup || newchan->_state != AST_STATE_UP || !transferer->_softhangup) {
ast_hangup(newchan);
if (f) {
ast_frfree(f);
@@ -792,17 +800,17 @@
return -1;
} else {
- ast_log(LOG_WARNING, "Unable to create channel Local/%s do you have chan_local?\n",dialstr);
ast_moh_stop(transferee);
ast_autoservice_stop(transferee);
ast_indicate(transferee, AST_CONTROL_UNHOLD);
- if (!ast_strlen_zero(xferfailsound)) {
+ /* any reason besides user requested cancel and busy triggers the failed sound */
+ if (outstate != AST_CONTROL_UNHOLD && outstate != AST_CONTROL_BUSY && !ast_strlen_zero(xferfailsound)) {
res = ast_streamfile(transferer, xferfailsound, transferer->language);
if (!res && (ast_waitstream(transferer, "") < 0)) {
return -1;
}
}
- return -1;
+ return FEATURE_RETURN_SUCCESS;
}
} else {
ast_log(LOG_WARNING, "Extension %s does not exist in context %s\n",xferto,transferer_real_context);
@@ -914,6 +922,178 @@
}
}
+
+static struct ast_channel *ast_feature_request_and_dial(struct ast_channel *caller, const char *type, int format, void *data, int timeout, int *outstate, const char *cid_num, const char *cid_name)
+{
+ int state = 0;
+ int cause = 0;
+ int to;
+ struct ast_channel *chan;
+ struct ast_channel *monitor_chans[2];
+ struct ast_channel *active_channel;
+ struct ast_frame *f = NULL;
+ int res = 0, ready = 0;
+
+ if ((chan = ast_request(type, format, data, &cause))) {
+ ast_set_callerid(chan, cid_num, cid_name, cid_num);
+
+ if (!ast_call(chan, data, timeout)) {
+ struct timeval started, ended;
+ int x, len = 0;
+ char *disconnect_code = NULL, *dialed_code = NULL;
+
+ ast_indicate(caller, AST_CONTROL_RINGING);
+ /* support dialing of the featuremap disconnect code while performing an attended tranfer */
+ for (x=0; x<FEATURES_COUNT; x++) {
+ if (strcasecmp(builtin_features[x].sname, "disconnect"))
+ continue;
+
+ disconnect_code = builtin_features[x].exten;
+ len = strlen(disconnect_code) + 1;
+ dialed_code = alloca(len);
+ memset(dialed_code, 0, len);
+ break;
+ }
+ x = 0;
+ gettimeofday(&started, NULL);
+ to = timeout;
+ while (!ast_check_hangup(caller) && timeout && (chan->_state != AST_STATE_UP)) {
+ monitor_chans[0] = caller;
+ monitor_chans[1] = chan;
+ active_channel = ast_waitfor_n(monitor_chans, 2, &to);
+
+ /* see if the timeout has been violated */
+ gettimeofday(&ended,NULL);
+ if(ast_tvdiff_ms(&started, &ended) > timeout) {
+ state = AST_CONTROL_UNHOLD;
+ ast_log(LOG_NOTICE, "We exceeded our AT-timeout\n");
+ break; /*doh! timeout*/
+ }
+
+ if (!active_channel) {
+ continue;
+ }
+
+ if (chan && (chan == active_channel)){
+ f = ast_read(chan);
+ if (f == NULL) { /*doh! where'd he go?*/
+ state = AST_CONTROL_HANGUP;
+ res = 0;
+ break;
+ }
+
+ if (f->frametype == AST_FRAME_CONTROL || f->frametype == AST_FRAME_DTMF || f->frametype == AST_FRAME_TEXT) {
+ if (f->subclass == AST_CONTROL_RINGING) {
+ state = f->subclass;
+ if (option_verbose > 2)
+ ast_verbose( VERBOSE_PREFIX_3 "%s is ringing\n", chan->name);
+ ast_indicate(caller, AST_CONTROL_RINGING);
+ } else if ((f->subclass == AST_CONTROL_BUSY) || (f->subclass == AST_CONTROL_CONGESTION)) {
+ state = f->subclass;
+ ast_frfree(f);
+ f = NULL;
+ break;
+ } else if (f->subclass == AST_CONTROL_ANSWER) {
+ /* This is what we are hoping for */
+ state = f->subclass;
+ ast_frfree(f);
+ f = NULL;
+ ready=1;
+ break;
+ } else {
+ ast_log(LOG_NOTICE, "Don't know what to do about control frame: %d\n", f->subclass);
+ }
+ /* else who cares */
+ }
+
+ } else if (caller && (active_channel == caller)) {
+ f = ast_read(caller);
+ if (f == NULL) { /*doh! where'd he go?*/
+ if (caller->_softhangup && !chan->_softhangup) {
+ /* make this a blind transfer */
+ ready = 1;
+ break;
+ }
+ state = AST_CONTROL_HANGUP;
+ res = 0;
+ break;
+ }
+
+ if (f->frametype == AST_FRAME_DTMF) {
+ dialed_code[x++] = f->subclass;
+ dialed_code[x] = '\0';
+ if (strlen(dialed_code) == len) {
+ x = 0;
+ } else if (x && strncmp(dialed_code, disconnect_code, x)) {
+ x = 0;
+ dialed_code[x] = '\0';
+ }
+ if (*dialed_code && !strcmp(dialed_code, disconnect_code)) {
+ /* Caller Canceled the call */
+ state = AST_CONTROL_UNHOLD;
+ ast_frfree(f);
+ f = NULL;
+ break;
+ }
+ }
+ }
+ if (f) {
+ ast_frfree(f);
+ }
+ }
+ } else
+ ast_log(LOG_NOTICE, "Unable to call channel %s/%s\n", type, (char *)data);
+ } else {
+ ast_log(LOG_NOTICE, "Unable to request channel %s/%s\n", type, (char *)data);
+ switch(cause) {
+ case AST_CAUSE_BUSY:
+ state = AST_CONTROL_BUSY;
+ break;
+ case AST_CAUSE_CONGESTION:
+ state = AST_CONTROL_CONGESTION;
+ break;
+ }
+ }
+
+ ast_indicate(caller, -1);
+ if (chan && ready) {
+ if (chan->_state == AST_STATE_UP)
+ state = AST_CONTROL_ANSWER;
+ res = 0;
+ } else if(chan) {
+ res = -1;
+ ast_hangup(chan);
+ chan = NULL;
+ } else {
+ res = -1;
+ }
+
+ if (outstate)
+ *outstate = state;
+
+ if (chan && res <= 0) {
+ if (!chan->cdr) {
+ chan->cdr = ast_cdr_alloc();
+ }
+ if (chan->cdr) {
+ char tmp[256];
+ ast_cdr_init(chan->cdr, chan);
+ snprintf(tmp, 256, "%s/%s", type, (char *)data);
+ ast_cdr_setapp(chan->cdr,"Dial",tmp);
+ ast_cdr_update(chan);
+ ast_cdr_start(chan->cdr);
+ ast_cdr_end(chan->cdr);
+ /* If the cause wasn't handled properly */
+ if (ast_cdr_disposition(chan->cdr,chan->hangupcause))
+ ast_cdr_failed(chan->cdr);
+ } else {
+ ast_log(LOG_WARNING, "Unable to create Call Detail Record\n");
+ }
+ }
+
+ return chan;
+}
+
int ast_bridge_call(struct ast_channel *chan,struct ast_channel *peer,struct ast_bridge_config *config)
{
/* Copy voice back and forth between the two channels. Give the peer
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