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asterisk/apps/app_dial.c

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/*
* Asterisk -- A telephony toolkit for Linux.
*
* Trivial application to dial a channel and send an URL on answer
*
* Copyright (C) 1999, Mark Spencer
*
* Mark Spencer <markster@linux-support.net>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*/
#include <asterisk/lock.h>
#include <asterisk/file.h>
#include <asterisk/logger.h>
#include <asterisk/channel.h>
#include <asterisk/pbx.h>
#include <asterisk/options.h>
#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>
#include <asterisk/utils.h>
#include <asterisk/app.h>
#include <stdlib.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <sys/time.h>
#include <sys/signal.h>
#include <netinet/in.h>
static char *tdesc = "Dialing Application";
static char *app = "Dial";
static char *synopsis = "Place a call and connect to the current channel";
static char *descrip =
" Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]):\n"
"Requests one or more channels and places specified outgoing calls on them.\n"
"As soon as a channel answers, the Dial app will answer the originating\n"
"channel (if it needs to be answered) and will bridge a call with the channel\n"
"which first answered. All other calls placed by the Dial app will be hung up\n"
"If a timeout is not specified, the Dial application will wait indefinitely\n"
"until either one of the called channels answers, the user hangs up, or all\n"
"channels return busy or error. In general, the dialer will return 0 if it\n"
"was unable to place the call, or the timeout expired. However, if all\n"
"channels were busy, and there exists an extension with priority n+101 (where\n"
"n is the priority of the dialer instance), then it will be the next\n"
"executed extension (this allows you to setup different behavior on busy from\n"
"no-answer).\n"
" This application returns -1 if the originating channel hangs up, or if the\n"
"call is bridged and either of the parties in the bridge terminate the call.\n"
"The option string may contain zero or more of the following characters:\n"
" 't' -- allow the called user transfer the calling user by hitting #.\n"
" 'T' -- allow the calling user to transfer the call by hitting #.\n"
" 'f' -- Forces callerid to be set as the extension of the line making/redirecting the outgoing call.\n"
" For example, some PSTNs don't allow callerids from other extensions then the ones\n"
" that are assigned to you.\n"
" 'r' -- indicate ringing to the calling party, pass no audio until answered.\n"
" 'm' -- provide hold music to the calling party until answered.\n"
" 'H' -- allow caller to hang up by hitting *.\n"
" 'C' -- reset call detail record for this call.\n"
" 'P[(x)]' -- privacy mode, using 'x' as database if provided.\n"
" '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"
" 'D([digits])' -- Send DTMF digit string *after* called party has answered\n"
" but before the bridge. (w=500ms sec pause)\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"
"it.\n";
/* We define a customer "local user" structure because we
use it not only for keeping track of what is in use but
also for keeping track of who we're dialing. */
struct localuser {
struct ast_channel *chan;
int stillgoing;
int allowredirect_in;
int allowredirect_out;
int ringbackonly;
int musiconhold;
int allowdisconnect;
int forcecallerid;
struct localuser *next;
};
LOCAL_USER_DECL;
static void hanguptree(struct localuser *outgoing, struct ast_channel *exception)
{
/* Hang up a tree of stuff */
struct localuser *oo;
while(outgoing) {
/* Hangup any existing lines we have open */
if (outgoing->chan && (outgoing->chan != exception))
ast_hangup(outgoing->chan);
oo = outgoing;
outgoing=outgoing->next;
free(oo);
}
}
#define AST_MAX_WATCHERS 256
static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localuser *outgoing, int *to, int *allowredir_in, int *allowredir_out, int *allowdisconnect, int *sentringing, char *status)
{
struct localuser *o;
int found;
int numlines;
int numbusy = 0;
int numcongestion = 0;
int numnochan = 0;
int orig = *to;
struct ast_frame *f;
struct ast_channel *peer = NULL;
struct ast_channel *watchers[AST_MAX_WATCHERS];
int pos;
int single;
struct ast_channel *winner;
single = (outgoing && !outgoing->next && !outgoing->musiconhold && !outgoing->ringbackonly);
if (single) {
/* Turn off hold music, etc */
ast_deactivate_generator(in);
/* If we are calling a single channel, make them compatible for in-band tone purpose */
ast_channel_make_compatible(outgoing->chan, in);
}
while(*to && !peer) {
o = outgoing;
found = -1;
pos = 1;
numlines = 0;
watchers[0] = in;
while(o) {
/* Keep track of important channels */
if (o->stillgoing && o->chan) {
watchers[pos++] = o->chan;
found = 1;
}
o = o->next;
numlines++;
}
if (found < 0) {
if (numlines == (numbusy + numcongestion + numnochan)) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_2 "Everyone is busy/congested at this time\n");
if (numbusy)
strcpy(status, "BUSY");
else if (numcongestion)
strcpy(status, "CONGESTION");
else if (numnochan)
strcpy(status, "CHANUNAVAIL");
/* See if there is a special busy message */
if (ast_exists_extension(in, in->context, in->exten, in->priority + 101, in->callerid))
in->priority+=100;
} else {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_2 "No one is available to answer at this time\n");
}
*to = 0;
return NULL;
}
winner = ast_waitfor_n(watchers, pos, to);
o = outgoing;
while(o) {
if (o->stillgoing && o->chan && (o->chan->_state == AST_STATE_UP)) {
if (!peer) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "%s answered %s\n", o->chan->name, in->name);
peer = o->chan;
*allowredir_in = o->allowredirect_in;
*allowredir_out = o->allowredirect_out;
*allowdisconnect = o->allowdisconnect;
}
} else if (o->chan && (o->chan == winner)) {
if (!ast_strlen_zero(o->chan->call_forward)) {
char tmpchan[256]="";
char *stuff;
char *tech;
strncpy(tmpchan, o->chan->call_forward, sizeof(tmpchan) - 1);
if ((stuff = strchr(tmpchan, '/'))) {
*stuff = '\0';
stuff++;
tech = tmpchan;
} else {
snprintf(tmpchan, sizeof(tmpchan), "%s@%s", o->chan->call_forward, o->chan->context);
stuff = tmpchan;
tech = "Local";
}
/* Before processing channel, go ahead and check for forwarding */
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, o->chan->name);
/* Setup parameters */
ast_hangup(o->chan);
o->chan = ast_request(tech, in->nativeformats, stuff);
if (!o->chan) {
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s'\n", tech, stuff);
o->stillgoing = 0;
numnochan++;
} else {
if (o->chan->callerid)
free(o->chan->callerid);
o->chan->callerid = NULL;
if (o->forcecallerid) {
char *newcid = NULL;
if (strlen(in->macroexten))
newcid = in->macroexten;
else
newcid = in->exten;
o->chan->callerid = strdup(newcid);
if (!o->chan->callerid)
ast_log(LOG_WARNING, "Out of memory\n");
} else {
if (in->callerid)
o->chan->callerid = strdup(in->callerid);
if (!o->chan->callerid)
ast_log(LOG_WARNING, "Out of memory\n");
}
if (in->ani) {
if (o->chan->ani)
free(o->chan->ani);
o->chan->ani = malloc(strlen(in->ani) + 1);
if (o->chan->ani)
strncpy(o->chan->ani, in->ani, strlen(in->ani) + 1);
else
ast_log(LOG_WARNING, "Out of memory\n");
}
if (o->chan->rdnis)
free(o->chan->rdnis);
if (!ast_strlen_zero(in->macroexten))
o->chan->rdnis = strdup(in->macroexten);
else
o->chan->rdnis = strdup(in->exten);
if (ast_call(o->chan, tmpchan, 0)) {
ast_log(LOG_NOTICE, "Failed to dial on local channel for call forward to '%s'\n", tmpchan);
o->stillgoing = 0;
ast_hangup(o->chan);
o->chan = NULL;
numnochan++;
}
}
continue;
}
f = ast_read(winner);
if (f) {
if (f->frametype == AST_FRAME_CONTROL) {
switch(f->subclass) {
case AST_CONTROL_ANSWER:
/* This is our guy if someone answered. */
if (!peer) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "%s answered %s\n", o->chan->name, in->name);
peer = o->chan;
*allowredir_in = o->allowredirect_in;
*allowredir_out = o->allowredirect_out;
*allowdisconnect = o->allowdisconnect;
}
break;
case AST_CONTROL_BUSY:
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "%s is busy\n", o->chan->name);
in->hangupcause = o->chan->hangupcause;
ast_hangup(o->chan);
o->chan = NULL;
o->stillgoing = 0;
if (in->cdr)
ast_cdr_busy(in->cdr);
numbusy++;
break;
case AST_CONTROL_CONGESTION:
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "%s is circuit-busy\n", o->chan->name);
in->hangupcause = o->chan->hangupcause;
ast_hangup(o->chan);
o->chan = NULL;
o->stillgoing = 0;
if (in->cdr)
ast_cdr_busy(in->cdr);
numcongestion++;
break;
case AST_CONTROL_RINGING:
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "%s is ringing\n", o->chan->name);
if (!(*sentringing) && !outgoing->musiconhold) {
ast_indicate(in, AST_CONTROL_RINGING);
(*sentringing)++;
}
break;
case AST_CONTROL_PROGRESS:
if (option_verbose > 2)
ast_verbose ( VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", o->chan->name,in->name);
ast_indicate(in, AST_CONTROL_PROGRESS);
break;
case AST_CONTROL_OFFHOOK:
/* Ignore going off hook */
break;
case -1:
if (!outgoing->ringbackonly && !outgoing->musiconhold) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "%s stopped sounds\n", o->chan->name);
ast_indicate(in, -1);
(*sentringing) = 0;
}
break;
default:
ast_log(LOG_DEBUG, "Dunno what to do with control type %d\n", f->subclass);
}
} else if (single && (f->frametype == AST_FRAME_VOICE) &&
!(outgoing->ringbackonly || outgoing->musiconhold)) {
if (ast_write(in, f))
ast_log(LOG_WARNING, "Unable to forward frame\n");
} else if (single && (f->frametype == AST_FRAME_IMAGE) &&
!(outgoing->ringbackonly || outgoing->musiconhold)) {
if (ast_write(in, f))
ast_log(LOG_WARNING, "Unable to forward image\n");
}
ast_frfree(f);
} else {
in->hangupcause = o->chan->hangupcause;
ast_hangup(o->chan);
o->chan = NULL;
o->stillgoing = 0;
}
}
o = o->next;
}
if (winner == in) {
f = ast_read(in);
#if 0
if (f && (f->frametype != AST_FRAME_VOICE))
printf("Frame type: %d, %d\n", f->frametype, f->subclass);
else if (!f || (f->frametype != AST_FRAME_VOICE))
printf("Hangup received on %s\n", in->name);
#endif
if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
/* Got hung up */
*to=-1;
strcpy(status, "CANCEL");
return NULL;
}
if (f && (f->frametype == AST_FRAME_DTMF) && *allowdisconnect &&
(f->subclass == '*')) {
if (option_verbose > 3)
ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
*to=0;
return NULL;
}
if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF))) {
if (ast_write(outgoing->chan, f))
ast_log(LOG_WARNING, "Unable to forward voice\n");
ast_frfree(f);
}
}
if (!*to && (option_verbose > 2))
ast_verbose( VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", orig);
}
return peer;
}
static int dial_exec(struct ast_channel *chan, void *data)
{
int res=-1;
struct localuser *u;
char info[256], *peers, *timeout, *tech, *number, *rest, *cur;
char privdb[256] = "", *s;
char announcemsg[256] = "", *ann;
struct localuser *outgoing=NULL, *tmp;
struct ast_channel *peer;
int to;
int allowredir_in=0;
int allowredir_out=0;
int allowdisconnect=0;
int privacy=0;
int announce=0;
int resetcdr=0;
char numsubst[AST_MAX_EXTENSION];
char restofit[AST_MAX_EXTENSION];
char *transfer = NULL;
char *newnum;
char callerid[256], *l, *n;
char *url=NULL; /* JDG */
struct ast_var_t *current;
struct varshead *headp, *newheadp;
struct ast_var_t *newvar;
int go_on=0;
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 *sdtmfptr;
char sdtmfdata[256] = "";
char *stack,*var;
char status[256];
int play_to_caller=0,play_to_callee=0;
int playargs=0, sentringing=0, moh=0;
int digit = 0;
if (!data) {
ast_log(LOG_WARNING, "Dial requires an argument (technology1/number1&technology2/number2...|optional timeout|options)\n");
return -1;
}
LOCAL_USER_ADD(u);
strncpy(info, (char *)data, sizeof(info) - 1);
peers = info;
if (peers) {
timeout = strchr(info, '|');
if (timeout) {
*timeout = '\0';
timeout++;
transfer = strchr(timeout, '|');
if (transfer) {
*transfer = '\0';
transfer++;
/* JDG */
url = strchr(transfer, '|');
if (url) {
*url = '\0';
url++;
ast_log(LOG_DEBUG, "DIAL WITH URL=%s_\n", url);
} else
ast_log(LOG_DEBUG, "SIMPLE DIAL (NO URL)\n");
/* /JDG */
}
}
} else
timeout = NULL;
if (!peers || ast_strlen_zero(peers)) {
ast_log(LOG_WARNING, "Dial argument takes format (technology1/number1&technology2/number2...|optional timeout)\n");
goto out;
}
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);
}
/* DTMF SCRIPT*/
if ((sdtmfptr = strstr(transfer, "D("))) {
strncpy(sdtmfdata, sdtmfptr + 2, sizeof(sdtmfdata) - 1);
/* Overwrite with X's what was the sdtmf info */
while (*sdtmfptr && (*sdtmfptr != ')'))
*(sdtmfptr++) = 'X';
if (*sdtmfptr)
*sdtmfptr = 'X';
/* Now find the end */
sdtmfptr = strchr(sdtmfdata, ')');
if (sdtmfptr)
*sdtmfptr = '\0';
else
ast_log(LOG_WARNING, "D( Data lacking trailing ')'\n");
}
/* 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 */
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;
}
/* undo effect of S(x) in case they are both used */
calldurationlimit=0;
/* more efficient do it like S(x) does since no advanced opts*/
if (!play_warning && !start_sound && !end_sound && timelimit) {
calldurationlimit=timelimit/1000;
timelimit=play_to_caller=play_to_callee=play_warning=warning_freq=0;
} else if (option_verbose > 2) {
ast_verbose(VERBOSE_PREFIX_3"Limit Data:\n");
ast_verbose(VERBOSE_PREFIX_3"timelimit=%ld\n",timelimit);
ast_verbose(VERBOSE_PREFIX_3"play_warning=%ld\n",play_warning);
ast_verbose(VERBOSE_PREFIX_3"play_to_caller=%s\n",play_to_caller ? "yes" : "no");
ast_verbose(VERBOSE_PREFIX_3"play_to_callee=%s\n",play_to_callee ? "yes" : "no");
ast_verbose(VERBOSE_PREFIX_3"warning_freq=%ld\n",warning_freq);
ast_verbose(VERBOSE_PREFIX_3"start_sound=%s\n",start_sound ? start_sound : "UNDEF");
ast_verbose(VERBOSE_PREFIX_3"warning_sound=%s\n",warning_sound ? warning_sound : "UNDEF");
ast_verbose(VERBOSE_PREFIX_3"end_sound=%s\n",end_sound ? end_sound : "UNDEF");
}
}
/* XXX ANNOUNCE SUPPORT */
if ((ann = strstr(transfer, "A("))) {
announce = 1;
strncpy(announcemsg, ann + 2, sizeof(announcemsg) - 1);
/* Overwrite with X's what was the announce info */
while(*ann && (*ann != ')'))
*(ann++) = 'X';
if (*ann)
*ann = 'X';
/* Now find the end of the privdb */
ann = strchr(announcemsg, ')');
if (ann)
*ann = '\0';
else {
ast_log(LOG_WARNING, "Transfer with Announce spec lacking trailing ')'\n");
announce = 0;
}
}
/* Extract privacy info from transfer */
if ((s = strstr(transfer, "P("))) {
privacy = 1;
strncpy(privdb, s + 2, sizeof(privdb) - 1);
/* Overwrite with X's what was the privacy info */
while(*s && (*s != ')'))
*(s++) = 'X';
if (*s)
*s = 'X';
/* Now find the end of the privdb */
s = strchr(privdb, ')');
if (s)
*s = '\0';
else {
ast_log(LOG_WARNING, "Transfer with privacy lacking trailing ')'\n");
privacy = 0;
}
} else if (strchr(transfer, 'P')) {
/* No specified privdb */
privacy = 1;
} else if (strchr(transfer, 'C')) {
resetcdr = 1;
}
}
if (resetcdr && chan->cdr)
ast_cdr_reset(chan->cdr, 0);
if (ast_strlen_zero(privdb) && privacy) {
/* If privdb is not specified and we are using privacy, copy from extension */
strncpy(privdb, chan->exten, sizeof(privdb) - 1);
}
if (privacy) {
if (chan->callerid)
strncpy(callerid, chan->callerid, sizeof(callerid));
else
strcpy(callerid, "");
ast_callerid_parse(callerid, &n, &l);
if (l) {
ast_shrink_phone_number(l);
} else
l = "";
ast_log(LOG_NOTICE, "Privacy DB is '%s', privacy is %d, clid is '%s'\n", privdb, privacy, l);
}
cur = peers;
do {
/* Remember where to start next time */
rest = strchr(cur, '&');
if (rest) {
*rest = 0;
rest++;
}
/* Get a technology/[device:]number pair */
tech = cur;
number = strchr(tech, '/');
if (!number) {
ast_log(LOG_WARNING, "Dial argument takes format (technology1/[device:]number1&technology2/[device:]number2...|optional timeout)\n");
goto out;
}
*number = '\0';
number++;
tmp = malloc(sizeof(struct localuser));
if (!tmp) {
ast_log(LOG_WARNING, "Out of memory\n");
goto out;
}
memset(tmp, 0, sizeof(struct localuser));
if (transfer) {
if (strchr(transfer, 't'))
tmp->allowredirect_in = 1;
else tmp->allowredirect_in = 0;
if (strchr(transfer, 'T'))
tmp->allowredirect_out = 1;
else tmp->allowredirect_out = 0;
if (strchr(transfer, 'r'))
tmp->ringbackonly = 1;
else tmp->ringbackonly = 0;
if (strchr(transfer, 'm'))
tmp->musiconhold = 1;
else tmp->musiconhold = 0;
if (strchr(transfer, 'H'))
allowdisconnect = tmp->allowdisconnect = 1;
else allowdisconnect = tmp->allowdisconnect = 0;
if(strchr(transfer, 'g'))
go_on=1;
if (strchr(transfer, 'f'))
tmp->forcecallerid = 1;
else tmp->forcecallerid = 0;
}
strncpy(numsubst, number, sizeof(numsubst)-1);
/* If we're dialing by extension, look at the extension to know what to dial */
if ((newnum = strstr(numsubst, "BYEXTENSION"))) {
strncpy(restofit, newnum + strlen("BYEXTENSION"), sizeof(restofit)-1);
snprintf(newnum, sizeof(numsubst) - (newnum - numsubst), "%s%s", chan->exten,restofit);
if (option_debug)
ast_log(LOG_DEBUG, "Dialing by extension %s\n", numsubst);
}
/* Request the peer */
tmp->chan = ast_request(tech, chan->nativeformats, numsubst);
if (!tmp->chan) {
/* If we can't, just go on to the next call */
ast_log(LOG_NOTICE, "Unable to create channel of type '%s'\n", tech);
if (chan->cdr)
ast_cdr_busy(chan->cdr);
free(tmp);
cur = rest;
continue;
}
if (!ast_strlen_zero(tmp->chan->call_forward)) {
char tmpchan[256]="";
char *stuff;
char *tech;
strncpy(tmpchan, tmp->chan->call_forward, sizeof(tmpchan) - 1);
if ((stuff = strchr(tmpchan, '/'))) {
*stuff = '\0';
stuff++;
tech = tmpchan;
} else {
snprintf(tmpchan, sizeof(tmpchan), "%s@%s", tmp->chan->call_forward, tmp->chan->context);
stuff = tmpchan;
tech = "Local";
}
/* Before processing channel, go ahead and check for forwarding */
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Forwarding %s to '%s/%s' (thanks to %s)\n", chan->name, tech, stuff, tmp->chan->name);
/* Setup parameters */
ast_hangup(tmp->chan);
tmp->chan = ast_request(tech, chan->nativeformats, stuff);
if (!tmp->chan) {
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s'\n", tech, stuff);
free(tmp);
cur = rest;
continue;
}
}
/* If creating a SIP channel, look for a variable called */
/* VXML_URL in the calling channel and copy it to the */
/* new channel. */
/* Check for ALERT_INFO in the SetVar list. This is for */
/* SIP distinctive ring as per the RFC. For Cisco 7960s, */
/* SetVar(ALERT_INFO=<x>) where x is an integer value 1-5. */
/* However, the RFC says it should be a URL. -km- */
headp=&chan->varshead;
AST_LIST_TRAVERSE(headp,current,entries) {
if (!strcasecmp(ast_var_name(current),"VXML_URL") ||
!strcasecmp(ast_var_name(current), "ALERT_INFO") ||
!strcasecmp(ast_var_name(current), "OSPTOKEN") ||
!strcasecmp(ast_var_name(current), "OSPHANDLE"))
{
newvar=ast_var_assign(ast_var_name(current),ast_var_value(current));
newheadp=&tmp->chan->varshead;
AST_LIST_INSERT_HEAD(newheadp,newvar,entries);
}
}
tmp->chan->appl = "AppDial";
tmp->chan->data = "(Outgoing Line)";
tmp->chan->whentohangup = 0;
if (tmp->chan->callerid)
free(tmp->chan->callerid);
if (tmp->chan->ani)
free(tmp->chan->ani);
if (chan->callerid)
tmp->chan->callerid = strdup(chan->callerid);
else
tmp->chan->callerid = NULL;
/* Copy language from incoming to outgoing */
strcpy(tmp->chan->language, chan->language);
if (ast_strlen_zero(tmp->chan->musicclass))
strncpy(tmp->chan->musicclass, chan->musicclass, sizeof(tmp->chan->musicclass) - 1);
if (chan->ani)
tmp->chan->ani = strdup(chan->ani);
else
tmp->chan->ani = NULL;
/* Pass hidecallerid setting */
tmp->chan->restrictcid = chan->restrictcid;
/* Pass callingpres setting */
tmp->chan->callingpres = chan->callingpres;
/* Presense of ADSI CPE on outgoing channel follows ours */
tmp->chan->adsicpe = chan->adsicpe;
/* pass the digital flag */
ast_dup_flag(tmp->chan, chan, AST_FLAG_DIGITAL);
/* Place the call, but don't wait on the answer */
res = ast_call(tmp->chan, numsubst, 0);
/* Save the info in cdr's that we called them */
if (chan->cdr)
ast_cdr_setdestchan(chan->cdr, tmp->chan->name);
/* check the results of ast_call */
if (res) {
/* Again, keep going even if there's an error */
if (option_debug)
ast_log(LOG_DEBUG, "ast call on peer returned %d\n", res);
else if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Couldn't call %s\n", numsubst);
ast_hangup(tmp->chan);
free(tmp);
cur = rest;
continue;
} else
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Called %s\n", numsubst);
/* Put them in the list of outgoing thingies... We're ready now.
XXX If we're forcibly removed, these outgoing calls won't get
hung up XXX */
tmp->stillgoing = -1;
tmp->next = outgoing;
outgoing = tmp;
/* If this line is up, don't try anybody else */
if (outgoing->chan->_state == AST_STATE_UP)
break;
cur = rest;
} while(cur);
if (timeout && !ast_strlen_zero(timeout)) {
to = atoi(timeout);
if (to > 0)
to *= 1000;
else
ast_log(LOG_WARNING, "Invalid timeout specified: '%s'\n", timeout);
} else
to = -1;
if (outgoing) {
/* Our status will at least be NOANSWER */
strcpy(status, "NOANSWER");
if (outgoing->musiconhold) {
moh=1;
ast_moh_start(chan, NULL);
} else if (outgoing->ringbackonly) {
ast_indicate(chan, AST_CONTROL_RINGING);
sentringing++;
}
} else
strcpy(status, "CHANUNAVAIL");
peer = wait_for_answer(chan, outgoing, &to, &allowredir_in, &allowredir_out, &allowdisconnect, &sentringing, status);
if (!peer) {
if (to)
/* Musta gotten hung up */
res = -1;
else
/* Nobody answered, next please? */
res=0;
goto out;
}
if (peer) {
#ifdef OSP_SUPPORT
/* Once call is answered, ditch the OSP Handle */
pbx_builtin_setvar_helper(chan, "OSPHANDLE", "");
#endif
strcpy(status, "ANSWER");
/* Ah ha! Someone answered within the desired timeframe. Of course after this
we will always return with -1 so that it is hung up properly after the
conversation. */
hanguptree(outgoing, peer);
outgoing = NULL;
/* If appropriate, log that we have a destination channel */
if (chan->cdr)
ast_cdr_setdestchan(chan->cdr, peer->name);
if (peer->name)
pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
if (numsubst)
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", numsubst);
/* JDG: sendurl */
if( url && !ast_strlen_zero(url) && ast_channel_supports_html(peer) ) {
ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url);
ast_channel_sendurl( peer, url );
} /* /JDG */
if (announce && announcemsg) {
// Start autoservice on the other chan
res = ast_autoservice_start(chan);
// Now Stream the File
if (!res)
res = ast_streamfile(peer,announcemsg,peer->language);
if (!res) {
digit = ast_waitstream(peer, AST_DIGIT_ANY);
}
// Ok, done. stop autoservice
res = ast_autoservice_stop(chan);
if (digit > 0 && !res)
res = ast_senddigit(chan, digit);
else
res = digit;
} else
res = 0;
if (!res) {
if (calldurationlimit > 0) {
time(&now);
chan->whentohangup = now + calldurationlimit;
}
if (!ast_strlen_zero(sdtmfdata))
res = ast_dtmf_stream(peer,chan,sdtmfdata,250);
}
if (!res) {
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;
if (moh) {
moh = 0;
ast_moh_stop(chan);
} else if (sentringing) {
sentringing = 0;
ast_indicate(chan, -1);
}
/* Be sure no generators are left on it */
ast_deactivate_generator(chan);
/* Make sure channels are compatible */
res = ast_channel_make_compatible(chan, peer);
if (res < 0) {
ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", chan->name, peer->name);
ast_hangup(peer);
return -1;
}
res = ast_bridge_call(chan,peer,&config);
} else
res = -1;
if (res != AST_PBX_NO_HANGUP_PEER) {
if (!chan->_softhangup)
chan->hangupcause = peer->hangupcause;
ast_hangup(peer);
}
}
out:
if (moh) {
moh = 0;
ast_moh_stop(chan);
} else if (sentringing) {
sentringing = 0;
ast_indicate(chan, -1);
}
hanguptree(outgoing, NULL);
pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
LOCAL_USER_REMOVE(u);
if((go_on>0) && (!chan->_softhangup))
res=0;
return res;
}
int unload_module(void)
{
STANDARD_HANGUP_LOCALUSERS;
return ast_unregister_application(app);
}
int load_module(void)
{
int res;
res = ast_register_application(app, dial_exec, synopsis, descrip);
return res;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}
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