git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
660 lines
20 KiB
C
Executable file
660 lines
20 KiB
C
Executable file
/*
|
|
* 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/parking.h>
|
|
#include <asterisk/musiconhold.h>
|
|
#include <asterisk/callerid.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>
|
|
|
|
#include <pthread.h>
|
|
|
|
static char *tdesc = "Dialing Application";
|
|
|
|
static char *app = "Dial";
|
|
|
|
static char *synopsis = "Place an 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 hunp 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 dialler 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 dialler 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\n"
|
|
" 'T' -- to allow the calling user to transfer the call.\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"
|
|
" 'd' -- data-quality (modem) call (minimum delay).\n"
|
|
" 'c' -- clear-channel data call (PRI-PRI only).\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"
|
|
" In addition to transferring the call, a call may be parked and then picked\n"
|
|
"up by another user.\n"
|
|
" The optionnal 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;
|
|
int ringbackonly;
|
|
int musiconhold;
|
|
int dataquality;
|
|
int allowdisconnect;
|
|
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 MAX 256
|
|
|
|
static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localuser *outgoing, int *to, int *allowredir, int *allowdisconnect)
|
|
{
|
|
struct localuser *o;
|
|
int found;
|
|
int numlines;
|
|
int sentringing = 0;
|
|
int numbusies = 0;
|
|
int orig = *to;
|
|
struct ast_frame *f;
|
|
struct ast_channel *peer = NULL;
|
|
struct ast_channel *watchers[MAX];
|
|
int pos;
|
|
int single;
|
|
int moh=0;
|
|
int ringind=0;
|
|
struct ast_channel *winner;
|
|
|
|
single = (outgoing && !outgoing->next && !outgoing->musiconhold && !outgoing->ringbackonly);
|
|
|
|
if (single) {
|
|
/* If we are calling a single channel, make them compatible for in-band tone purpose */
|
|
ast_channel_make_compatible(outgoing->chan, in);
|
|
}
|
|
|
|
if (outgoing) {
|
|
moh = outgoing->musiconhold;
|
|
ringind = outgoing->ringbackonly;
|
|
if (outgoing->musiconhold) {
|
|
ast_moh_start(in, NULL);
|
|
} else if (outgoing->ringbackonly) {
|
|
ast_indicate(in, AST_CONTROL_RINGING);
|
|
}
|
|
}
|
|
|
|
while(*to && !peer) {
|
|
o = outgoing;
|
|
found = -1;
|
|
pos = 1;
|
|
numlines = 0;
|
|
watchers[0] = in;
|
|
while(o) {
|
|
/* Keep track of important channels */
|
|
if (o->stillgoing) {
|
|
watchers[pos++] = o->chan;
|
|
found = 1;
|
|
}
|
|
o = o->next;
|
|
numlines++;
|
|
}
|
|
if (found < 0) {
|
|
if (numlines == numbusies) {
|
|
if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_2 "Everyone is busy at this time\n");
|
|
/* 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;
|
|
/* if no one available we'd better stop MOH/ringing to */
|
|
if (moh) {
|
|
ast_moh_stop(in);
|
|
} else if (ringind) {
|
|
ast_indicate(in, -1);
|
|
}
|
|
return NULL;
|
|
}
|
|
winner = ast_waitfor_n(watchers, pos, to);
|
|
o = outgoing;
|
|
while(o) {
|
|
if (o->stillgoing && (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 = o->allowredirect;
|
|
*allowdisconnect = o->allowdisconnect;
|
|
}
|
|
} else if (o->chan == winner) {
|
|
if (strlen(o->chan->call_forward)) {
|
|
char tmpchan[256];
|
|
/* 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, o->chan->call_forward, o->chan->context, o->chan->name);
|
|
/* Setup parameters */
|
|
snprintf(tmpchan, sizeof(tmpchan),"%s@%s", o->chan->call_forward, o->chan->context);
|
|
ast_hangup(o->chan);
|
|
o->chan = ast_request("Local", in->nativeformats, tmpchan);
|
|
if (!o->chan) {
|
|
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s'\n", tmpchan);
|
|
o->stillgoing = 0;
|
|
numbusies++;
|
|
}
|
|
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;
|
|
}
|
|
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 = o->allowredirect;
|
|
*allowdisconnect = o->allowdisconnect;
|
|
}
|
|
break;
|
|
case AST_CONTROL_BUSY:
|
|
if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_3 "%s is busy\n", o->chan->name);
|
|
o->stillgoing = 0;
|
|
if (in->cdr)
|
|
ast_cdr_busy(in->cdr);
|
|
numbusies++;
|
|
break;
|
|
case AST_CONTROL_CONGESTION:
|
|
if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_3 "%s is circuit-busy\n", o->chan->name);
|
|
o->stillgoing = 0;
|
|
if (in->cdr)
|
|
ast_cdr_busy(in->cdr);
|
|
numbusies++;
|
|
break;
|
|
case AST_CONTROL_RINGING:
|
|
if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_3 "%s is ringing\n", o->chan->name);
|
|
if (!sentringing) {
|
|
ast_indicate(in, AST_CONTROL_RINGING);
|
|
sentringing++;
|
|
ringind++;
|
|
}
|
|
break;
|
|
case AST_CONTROL_OFFHOOK:
|
|
/* Ignore going off hook */
|
|
break;
|
|
case -1:
|
|
if (option_verbose > 2)
|
|
ast_verbose( VERBOSE_PREFIX_3 "%s stopped sounds\n", o->chan->name);
|
|
ast_indicate(in, -1);
|
|
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 {
|
|
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;
|
|
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");
|
|
}
|
|
}
|
|
if (!*to && (option_verbose > 2))
|
|
ast_verbose( VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", orig);
|
|
}
|
|
if (moh) {
|
|
ast_moh_stop(in);
|
|
} else if (ringind) {
|
|
ast_indicate(in, -1);
|
|
}
|
|
|
|
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;
|
|
struct localuser *outgoing=NULL, *tmp;
|
|
struct ast_channel *peer;
|
|
int to;
|
|
int allowredir=0;
|
|
int allowdisconnect=0;
|
|
int privacy=0;
|
|
int resetcdr=0;
|
|
int clearchannel=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;
|
|
|
|
if (!data) {
|
|
ast_log(LOG_WARNING, "Dial requires an argument (technology1/number1&technology2/number2...|optional timeout)\n");
|
|
return -1;
|
|
}
|
|
|
|
LOCAL_USER_ADD(u);
|
|
|
|
/* Parse our arguments XXX Check for failure XXX */
|
|
strncpy(info, (char *)data, strlen((char *)data) + AST_MAX_EXTENSION-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 || !strlen(peers)) {
|
|
ast_log(LOG_WARNING, "Dial argument takes format (technology1/number1&technology2/number2...|optional timeout)\n");
|
|
goto out;
|
|
}
|
|
|
|
|
|
if (transfer) {
|
|
/* 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 (!strlen(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 = 1;
|
|
else tmp->allowredirect = 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, 'd'))
|
|
tmp->dataquality = 1;
|
|
else tmp->dataquality = 0;
|
|
if (strchr(transfer, 'H'))
|
|
tmp->allowdisconnect = 1;
|
|
else tmp->allowdisconnect = 0;
|
|
if (strchr(transfer, 'c'))
|
|
clearchannel = 1;
|
|
else
|
|
clearchannel = 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 (strlen(tmp->chan->call_forward)) {
|
|
char tmpchan[256];
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Forwarding call to '%s@%s'\n", tmp->chan->call_forward, tmp->chan->context);
|
|
snprintf(tmpchan, sizeof(tmpchan),"%s@%s", tmp->chan->call_forward, tmp->chan->context);
|
|
ast_hangup(tmp->chan);
|
|
tmp->chan = ast_request("Local", chan->nativeformats, tmpchan);
|
|
if (!tmp->chan) {
|
|
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s'\n", tmpchan);
|
|
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. */
|
|
if (strcasecmp(tech,"SIP")==0)
|
|
{
|
|
headp=&chan->varshead;
|
|
AST_LIST_TRAVERSE(headp,current,entries) {
|
|
if (strcasecmp(ast_var_name(current),"VXML_URL")==0)
|
|
{
|
|
newvar=ast_var_assign(ast_var_name(current),ast_var_value(current));
|
|
newheadp=&tmp->chan->varshead;
|
|
AST_LIST_INSERT_HEAD(newheadp,newvar,entries);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
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;
|
|
if (chan->ani)
|
|
tmp->chan->ani = strdup(chan->ani);
|
|
else
|
|
tmp->chan->ani = NULL;
|
|
/* Presense of ADSI CPE on outgoing channel follows ours */
|
|
tmp->chan->adsicpe = chan->adsicpe;
|
|
/* Place the call, but don't wait on the answer */
|
|
res = ast_call(tmp->chan, numsubst, 0);
|
|
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 && strlen(timeout))
|
|
to = atoi(timeout) * 1000;
|
|
else
|
|
to = -1;
|
|
peer = wait_for_answer(chan, outgoing, &to, &allowredir, &allowdisconnect);
|
|
if (!peer) {
|
|
if (to)
|
|
/* Musta gotten hung up */
|
|
res = -1;
|
|
else
|
|
/* Nobody answered, next please? */
|
|
res=0;
|
|
|
|
goto out;
|
|
}
|
|
if (peer) {
|
|
/* 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. */
|
|
if (!strcmp(chan->type,"Zap"))
|
|
{
|
|
int x = 2;
|
|
if (tmp->dataquality || clearchannel) x = 0;
|
|
ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
|
|
}
|
|
if (!strcmp(peer->type,"Zap"))
|
|
{
|
|
int x = 2;
|
|
if (tmp->dataquality || clearchannel) x = 0;
|
|
ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
|
|
}
|
|
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);
|
|
/* 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;
|
|
}
|
|
/* JDG: sendurl */
|
|
if( url && strlen(url) && ast_channel_supports_html(peer) ) {
|
|
ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url);
|
|
ast_channel_sendurl( peer, url );
|
|
} /* /JDG */
|
|
if (clearchannel)
|
|
{
|
|
int x = 0;
|
|
ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
|
|
ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
|
|
}
|
|
res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | clearchannel);
|
|
if (clearchannel)
|
|
{
|
|
int x = 1;
|
|
ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
|
|
ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
|
|
}
|
|
ast_hangup(peer);
|
|
}
|
|
out:
|
|
hanguptree(outgoing, NULL);
|
|
LOCAL_USER_REMOVE(u);
|
|
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;
|
|
}
|