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applied final release of bug 1353 per Mark's permission

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Anthony Minessale II
2004-04-26 23:22:34 +00:00
parent f4b55b802c
commit 5385ca0a0e
7 changed files with 447 additions and 71 deletions
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109
apps/app_dial.c
View File

@@ -20,6 +20,7 @@
#include <asterisk/module.h>
#include <asterisk/translate.h>
#include <asterisk/say.h>
#include <asterisk/config.h>
#include <asterisk/parking.h>
#include <asterisk/musiconhold.h>
#include <asterisk/callerid.h>
@@ -68,6 +69,15 @@ static char *descrip =
" 'g' -- goes on in context if the destination channel hangs up\n"
" 'A(x)' -- play an announcement to the called party, using x as file\n"
" 'S(x)' -- hangup the call after x seconds AFTER called party picked up\n"
" 'L(x[:y][:z])' -- Limit the call to 'x' ms warning when 'y' ms are left (repeated every 'z' ms)\n"
" -- Only 'x' is required, 'y' and 'z' are optional.\n"
" -- The following special variables are optional:\n"
" ** LIMIT_PLAYAUDIO_CALLER (default yes) Play sounds to the caller.\n"
" ** LIMIT_PLAYAUDIO_CALLEE Play sounds to the callee.\n"
" ** LIMIT_TIMEOUT_FILE File to play when time is up.\n"
" ** LIMIT_CONNECT_FILE File to play when call begins.\n"
" ** LIMIT_WARNING_FILE File to play as warning if 'y' is defined.\n"
" -- 'timeleft' is a special sound macro to auto-say the time left and is the default.\n\n"
" In addition to transferring the call, a call may be parked and then picked\n"
"up by another user.\n"
" The optional URL will be sent to the called party if the channel supports\n"
@@ -390,7 +400,19 @@ static int dial_exec(struct ast_channel *chan, void *data)
unsigned int calldurationlimit=0;
char *cdl;
time_t now;
struct ast_bridge_config config;
long timelimit = 0;
long play_warning = 0;
long warning_freq=0;
char *warning_sound=NULL;
char *end_sound=NULL;
char *start_sound=NULL;
char *limitptr;
char limitdata[256];
char *stack,*var;
int play_to_caller=0,play_to_callee=0;
int playargs=0;
if (!data) {
ast_log(LOG_WARNING, "Dial requires an argument (technology1/number1&technology2/number2...|optional timeout|options)\n");
return -1;
@@ -430,13 +452,81 @@ static int dial_exec(struct ast_channel *chan, void *data)
if (transfer) {
/* Extract call duration limit */
if ((cdl = strstr(transfer, "S("))) {
calldurationlimit=atoi(cdl+2);
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Setting call duration limit to %i seconds.\n",calldurationlimit);
}
/* XXX LIMIT SUPPORT */
if ((limitptr = strstr(transfer, "L("))) {
strncpy(limitdata, limitptr + 2, sizeof(limitdata) - 1);
/* Overwrite with X's what was the limit info */
while(*limitptr && (*limitptr != ')'))
*(limitptr++) = 'X';
if (*limitptr)
*limitptr = 'X';
/* Now find the end of the privdb */
limitptr = strchr(limitdata, ')');
if (limitptr)
*limitptr = '\0';
else {
ast_log(LOG_WARNING, "Limit Data lacking trailing ')'\n");
}
var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLER");
play_to_caller = var ? ast_true(var) : 1;
var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLEE");
play_to_callee = var ? ast_true(var) : 0;
if(! play_to_caller && ! play_to_callee)
play_to_caller=1;
var = pbx_builtin_getvar_helper(chan,"LIMIT_WARNING_FILE");
warning_sound = var ? var : "timeleft";
var = pbx_builtin_getvar_helper(chan,"LIMIT_TIMEOUT_FILE");
end_sound = var ? var : NULL;
var = pbx_builtin_getvar_helper(chan,"LIMIT_CONNECT_FILE");
start_sound = var ? var : NULL;
var=stack=limitdata;
var = strsep(&stack, ":");
if(var) {
timelimit = atol(var);
playargs++;
}
var = strsep(&stack, ":");
if(var) {
play_warning = atol(var);
playargs++;
}
var = strsep(&stack, ":");
if(var) {
warning_freq = atol(var);
playargs++;
}
if(! timelimit) {
timelimit=play_to_caller=play_to_callee=play_warning=warning_freq=0;
warning_sound=NULL;
}
calldurationlimit=0; /* undo effect of S(x) in case they are both used */
if(! play_warning && ! start_sound && ! end_sound && timelimit) { /* more efficient do it like S(x) does since no advanced opts*/
calldurationlimit=timelimit/1000;
timelimit=play_to_caller=play_to_callee=play_warning=warning_freq=0;
}
else
ast_verbose(VERBOSE_PREFIX_3"Limit Data: timelimit=%ld\n -- play_warning=%ld\n -- play_to_caller=%s\n -- play_to_callee=%s\n -- warning_freq=%ld\n -- warning_sound=%s\n -- end_sound=%s\n -- start_sound=%s\n",timelimit,play_warning,play_to_caller ? "yes" : "no",play_to_callee ? "yes" : "no",warning_freq,warning_sound ? warning_sound : "UNDEF",end_sound ? end_sound : "UNDEF",start_sound ? start_sound : "UNDEF");
}
/* XXX ANNOUNCE SUPPORT */
if ((ann = strstr(transfer, "A("))) {
announce = 1;
@@ -729,7 +819,20 @@ static int dial_exec(struct ast_channel *chan, void *data)
time(&now);
chan->whentohangup = now + calldurationlimit;
}
res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out, allowdisconnect);
memset(&config,0,sizeof(struct ast_bridge_config));
config.play_to_caller=play_to_caller;
config.play_to_callee=play_to_callee;
config.allowredirect_in = allowredir_in;
config.allowredirect_out = allowredir_out;
config.allowdisconnect = allowdisconnect;
config.timelimit = timelimit;
config.play_warning = play_warning;
config.warning_freq = warning_freq;
config.warning_sound = warning_sound;
config.end_sound = end_sound;
config.start_sound = start_sound;
res = ast_bridge_call(chan,peer,&config);
if (res != AST_PBX_NO_HANGUP_PEER)
ast_hangup(peer);

9
apps/app_queue.c
View File

@@ -874,6 +874,7 @@ static int try_calling(struct queue_ent *qe, char *options, char *announceoverri
char digit = 0;
time_t callstart;
time_t now;
struct ast_bridge_config config;
/* Hold the lock while we setup the outgoing calls */
ast_mutex_lock(&qe->parent->lock);
strncpy(queuename, qe->parent->name, sizeof(queuename) - 1);
@@ -1030,7 +1031,13 @@ static int try_calling(struct queue_ent *qe, char *options, char *announceoverri
strncpy(oldcontext, qe->chan->context, sizeof(oldcontext) - 1);
strncpy(oldexten, qe->chan->exten, sizeof(oldexten) - 1);
time(&callstart);
bridge = ast_bridge_call(qe->chan, peer, allowredir_in, allowredir_out, allowdisconnect);
memset(&config,0,sizeof(struct ast_bridge_config));
config.allowredirect_in = allowredir_in;
config.allowredirect_out = allowredir_out;
config.allowdisconnect = allowdisconnect;
bridge = ast_bridge_call(qe->chan,peer,&config);
if (strcasecmp(oldcontext, qe->chan->context) || strcasecmp(oldexten, qe->chan->exten)) {
ast_queue_log(queuename, qe->chan->uniqueid, peer->name, "TRANSFER", "%s|%s", qe->chan->exten, qe->chan->context);
} else if (qe->chan->_softhangup) {