diff --git a/CREDITS b/CREDITS
index cef7cfc1af096fc1151b24293f3e86eb574d315c..b971259ef0124875990661069b8d527cc6aaeb4c 100755
--- a/CREDITS
+++ b/CREDITS
@@ -6,6 +6,13 @@ and TSU 120e to the project. (http://www.adtran.com)
* Thanks to QuickNet Technologies for their donation of an Internet
PhoneJack card to the project. (http://www.quicknet.net)
+=== DEVELOPMENT SUPPORT ===
+I'd like to thank the following companies for helping fund development of
+Asterisk:
+
+* Celera Networks - US Digital
+* Adtran, Inc.
+
=== OTHER SOURCE CODE IN ASTERISK ===
I did not implement the codecs in asterisk. Here is the copyright on the
diff --git a/README b/README
index 0656a898500f4bee1bd118867d3386b442f7d305..704e31d6755fdc5591cb4830c116f0577c487401 100755
--- a/README
+++ b/README
@@ -1,7 +1,11 @@
The Asterisk Open Source PBX
by Mark Spencer <markster@linux-support.net>
-Copyright (C) 1999, Mark Spencer
+Copyright (C) 2001, Linux Support Services, Inc.
================================================================
+* SECURITY
+ It is imperative that you read and fully understand the contents of
+ the SECURITY file before you attempt to configure an Asterisk server.
+
* WHAT IS ASTERISK
Asterisk is an Open Source PBX and telephony toolkit. It is, in a
sense, middleware between Internet and telephony channels on the bottom,
@@ -10,7 +14,27 @@ on the project itself, please visit the Asterisk home page at:
http://www.asteriskpbx.com
-
+* LICENSING
+ Asterisk is distributed under GNU General Public License. The GPL also
+must apply to all loadable modules as well, except as defined below.
+
+ Linux Support Services, Inc. retains copyright to all of the core
+Asterisk system, and therefore can grant, at its sole discression, the
+ability for companies, individuals, or organizations to create proprietary
+or Open Source (but non-GPL'd) modules which may be dynamically linked at
+runtime with the portions of Asterisk which fall under our copyright
+umbrella, or are distributed under more flexible licenses than GPL. At
+this time (5/21/2001) the only component of Asterisk which is covered
+under GPL and not under our Copyright is the Xing MP3 decoder.
+
+ If you wish to use our code in other GPL programs, don't worry -- there
+is no requirement that you provide the same exemption in your GPL'd
+products (although if you've written a module for Asterisk we would
+strongly encourage you to make the same excemption that we do).
+
+ If you have any questions, whatsoever, regarding our licensing policy,
+please contact us.
+
* REQUIRED COMPONENTS
== Linux ==
diff --git a/SECURITY b/SECURITY
new file mode 100755
index 0000000000000000000000000000000000000000..fd9873958127e19a0763efa3b40952d627ecf98c
--- /dev/null
+++ b/SECURITY
@@ -0,0 +1,38 @@
+==== Security Notes with Asterisk ====
+
+PLEASE READ THE FOLLOWING IMPORTANT SECURITY RELATED INFORMATION.
+IMPROPER CONFIGURATION OF ASTERISK COULD ALLOW UNAUTHORIZED USE OF YOUR
+FACILITIES, POTENTIALLY INCURRING SUBSTANTIAL CHARGES.
+
+First and foremost remember this:
+
+USE THE EXTENSION CONTEXTS TO ISOLATE OUTGOING OR TOLL SERVICES FROM ANY
+INCOMING CONNECTIONS.
+
+You should consider that if any channel, incoming line, etc can enter an
+extension context that it has the capability of accessing any extension
+within that context.
+
+Therefore, you should NOT allow access to outgoing or toll services in
+contexts that are accessible (especially without a password) from incoming
+channels, be they IAX channels, FX or other trunks, or even untrusted
+stations within you network. In particular, never ever put outgoing toll
+services in the "default" context. To make things easier, you can include
+the "default" context within other private contexts by using:
+
+ include => default
+
+in the appropriate section. A well designed PBX might look like this:
+
+[longdistance]
+exten => _91NXXNXXXXXX,1,Dial,Tor/g2/BYEXTENSION
+include => local
+
+[local]
+exten => _9NXXNXXX,1,Dial,Tor/g2/BYEXTENSION
+include => default
+
+[default]
+exten => 6123,Dial,Tor/1
+
+
diff --git a/apps/app_image.c b/apps/app_image.c
new file mode 100755
index 0000000000000000000000000000000000000000..cc677ff5c352fbe49de09049b75c865f10831551
--- /dev/null
+++ b/apps/app_image.c
@@ -0,0 +1,89 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * App to transmit an image
+ *
+ * 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/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/image.h>
+#include <string.h>
+#include <stdlib.h>
+#include <pthread.h>
+
+static char *tdesc = "Image Transmission Application";
+
+static char *app = "SendImage";
+
+static char *synopsis = "Send an image file";
+
+static char *descrip =
+" SendImage(filename): Sends an image on a channel. If the channel\n"
+"does not support image transport, and there exists a step with\n"
+"priority n + 101, then execution will continue at that step.\n"
+"Otherwise, execution will continue at the next priority level.\n"
+"SendImage only returns 0 if the image was sent correctly or if\n"
+"the channel does not support image transport, and -1 otherwise.\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int sendimage_exec(struct ast_channel *chan, void *data)
+{
+ int res = 0;
+ struct localuser *u;
+ if (!data || !strlen((char *)data)) {
+ ast_log(LOG_WARNING, "SendImage requires an argument (filename)\n");
+ return -1;
+ }
+ LOCAL_USER_ADD(u);
+ if (!ast_supports_images(chan)) {
+ /* Does not support transport */
+ if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+ chan->priority += 100;
+ return 0;
+ }
+ res = ast_send_image(chan, data);
+ LOCAL_USER_REMOVE(u);
+ return res;
+}
+
+int unload_module(void)
+{
+ STANDARD_HANGUP_LOCALUSERS;
+ return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+ return ast_register_application(app, sendimage_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+ return tdesc;
+}
+
+int usecount(void)
+{
+ int res;
+ STANDARD_USECOUNT(res);
+ return res;
+}
+
+char *key()
+{
+ return ASTERISK_GPL_KEY;
+}
diff --git a/apps/app_system.c b/apps/app_system.c
index bb10543e244f9799783f53ff3a4bd3ae723276c0..a4369125f2213b0794f1614973b6869192b23430 100755
--- a/apps/app_system.c
+++ b/apps/app_system.c
@@ -31,10 +31,11 @@ static char *app = "System";
static char *synopsis = "Execute a system command";
static char *descrip =
-" System(command): Executes a command by using system(). Returns -1 on failure to execute\n"
-" the specified command. If the command itself executes but is in error, and if there exists\n"
-" a priority n + 101, where 'n' is the priority of the current instance, then the channel will\n"
-" will be setup to continue at that priority level. Otherwise, System returns 0.\n";
+" System(command): Executes a command by using system(). Returns -1 on\n"
+"failure to execute the specified command. If the command itself executes\n"
+"but is in error, and if there exists a priority n + 101, where 'n' is the\n"
+"priority of the current instance, then the channel will be setup to\n"
+"continue at that priority level. Otherwise, System returns 0.\n";
STANDARD_LOCAL_USER;
@@ -58,7 +59,7 @@ static int skel_exec(struct ast_channel *chan, void *data)
ast_log(LOG_WARNING, "Unable to execute '%s'\n", data);
res = -1;
} else {
- if (res && ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101))
+ if (res && ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
chan->priority+=100;
res = 0;
}
diff --git a/apps/app_url.c b/apps/app_url.c
new file mode 100755
index 0000000000000000000000000000000000000000..542c490a789c7685543e3cab46ee30cf486073e8
--- /dev/null
+++ b/apps/app_url.c
@@ -0,0 +1,137 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * App to transmit a URL
+ *
+ * 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/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/image.h>
+#include <string.h>
+#include <stdlib.h>
+#include <pthread.h>
+
+static char *tdesc = "Send URL Applications";
+
+static char *app = "SendURL";
+
+static char *synopsis = "Send a URL";
+
+static char *descrip =
+" SendURL(URL[|option]): Requests client go to URL. If the client\n"
+"does not support html transport, and there exists a step with\n"
+"priority n + 101, then execution will continue at that step.\n"
+"Otherwise, execution will continue at the next priority level.\n"
+"SendURL only returns 0 if the URL was sent correctly or if\n"
+"the channel does not support HTML transport, and -1 otherwise.\n"
+"If the option 'wait' is specified, execution will wait for an\n"
+"acknowledgement that the URL has been loaded before continuing\n"
+"and will return -1 if the peer is unable to load the URL\n";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int sendurl_exec(struct ast_channel *chan, void *data)
+{
+ int res = 0;
+ struct localuser *u;
+ char tmp[256];
+ char *options;
+ int option_wait=0;
+ struct ast_frame *f;
+ if (!data || !strlen((char *)data)) {
+ ast_log(LOG_WARNING, "SendURL requires an argument (URL)\n");
+ return -1;
+ }
+ strncpy(tmp, (char *)data, sizeof(tmp));
+ strtok(tmp, "|");
+ options = strtok(NULL, "|");
+ if (options && !strcasecmp(options, "wait"))
+ option_wait = 1;
+ LOCAL_USER_ADD(u);
+ if (!ast_channel_supports_html(chan)) {
+ /* Does not support transport */
+ if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+ chan->priority += 100;
+ LOCAL_USER_REMOVE(u);
+ return 0;
+ }
+ res = ast_channel_sendurl(chan, tmp);
+ if (res > -1) {
+ if (option_wait) {
+ for(;;) {
+ /* Wait for an event */
+ res = ast_waitfor(chan, -1);
+ if (res < 0)
+ break;
+ f = ast_read(chan);
+ if (!f) {
+ res = -1;
+ break;
+ }
+ if (f->frametype == AST_FRAME_HTML) {
+ switch(f->subclass) {
+ case AST_HTML_LDCOMPLETE:
+ res = 0;
+ ast_frfree(f);
+ goto out;
+ break;
+ case AST_HTML_NOSUPPORT:
+ /* Does not support transport */
+ if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101, chan->callerid))
+ chan->priority += 100;
+ res = 0;
+ goto out;
+ break;
+ default:
+ ast_log(LOG_WARNING, "Don't know what to do with HTML subclass %d\n", f->subclass);
+ };
+ }
+ ast_frfree(f);
+ }
+ }
+ }
+out:
+ LOCAL_USER_REMOVE(u);
+ return res;
+}
+
+int unload_module(void)
+{
+ STANDARD_HANGUP_LOCALUSERS;
+ return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+ return ast_register_application(app, sendurl_exec, synopsis, descrip);
+}
+
+char *description(void)
+{
+ return tdesc;
+}
+
+int usecount(void)
+{
+ int res;
+ STANDARD_USECOUNT(res);
+ return res;
+}
+
+char *key()
+{
+ return ASTERISK_GPL_KEY;
+}
diff --git a/callerid.c b/callerid.c
index f78714fbf45ab1b574c52292296b7446b81953d1..438d2b38b2dd5e0414810ae10c939ea79a3f92b9 100755
--- a/callerid.c
+++ b/callerid.c
@@ -62,7 +62,7 @@ struct callerid_state *callerid_new(void)
{
struct callerid_state *cid;
cid = malloc(sizeof(struct callerid_state));
- memset(cid, 0, sizeof(*cid));
+ memset(cid, 0, sizeof(struct callerid_state));
if (cid) {
cid->fskd.spb = 7; /* 1200 baud */
cid->fskd.hdlc = 0; /* Async */
@@ -146,6 +146,10 @@ int callerid_feed(struct callerid_state *cid, unsigned char *ubuf, int len)
while(mylen >= 80) {
olen = mylen;
res = fsk_serie(&cid->fskd, buf, &mylen, &b);
+ if (mylen < 0) {
+ ast_log(LOG_ERROR, "fsk_serie made mylen < 0 (%d)\n", mylen);
+ return -1;
+ }
buf += (olen - mylen);
if (res < 0) {
ast_log(LOG_NOTICE, "fsk_serie failed\n");
@@ -263,7 +267,8 @@ int callerid_feed(struct callerid_state *cid, unsigned char *ubuf, int len)
if (mylen) {
memcpy(cid->oldstuff, buf, mylen * 2);
cid->oldlen = mylen * 2;
- }
+ } else
+ cid->oldlen = 0;
free(obuf);
return 0;
}
@@ -444,7 +449,7 @@ void ast_shrink_phone_number(char *n)
int ast_isphonenumber(char *n)
{
int x;
- if (!n)
+ if (!n || !strlen(n))
return 0;
for (x=0;n[x];x++)
if (!strchr("0123456789", n[x]))
@@ -483,7 +488,7 @@ int ast_callerid_parse(char *instr, char **name, char **location)
} else {
strncpy(tmp, instr, sizeof(tmp));
ast_shrink_phone_number(tmp);
- if (!ast_isphonenumber(tmp)) {
+ if (ast_isphonenumber(tmp)) {
/* Assume it's just a location */
*name = NULL;
*location = instr;
diff --git a/channels/chan_oss.c b/channels/chan_oss.c
index 784067a9797422cc3eb974dc5f41e2345226920c..fab41ed14b0f0fbac5a4c856a7435aec1764a3e8 100755
--- a/channels/chan_oss.c
+++ b/channels/chan_oss.c
@@ -56,6 +56,7 @@ static struct timeval lasttime;
static int usecnt;
static int needanswer = 0;
+static int needringing = 0;
static int needhangup = 0;
static int silencesuppression = 0;
static int silencethreshold = 1000;
@@ -438,6 +439,7 @@ static int oss_call(struct ast_channel *c, char *dest, int timeout)
needanswer = 1;
} else {
ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
+ needringing = 1;
write(sndcmd[1], &res, sizeof(res));
}
return 0;
@@ -591,7 +593,15 @@ static struct ast_frame *oss_read(struct ast_channel *chan)
f.src = type;
f.mallocd = 0;
+ if (needringing) {
+ f.frametype = AST_FRAME_CONTROL;
+ f.subclass = AST_CONTROL_RINGING;
+ needringing = 0;
+ return &f;
+ }
+
if (needhangup) {
+ needhangup = 0;
return NULL;
}
if (strlen(text2send)) {
@@ -632,8 +642,10 @@ static struct ast_frame *oss_read(struct ast_channel *chan)
}
res = read(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
if (res < 0) {
- ast_log(LOG_WARNING, "Error reading from sound device: %s\n", strerror(errno));
+ ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno));
+#if 0
CRASH;
+#endif
return NULL;
}
readpos += res;
@@ -641,6 +653,10 @@ static struct ast_frame *oss_read(struct ast_channel *chan)
if (readpos >= FRAME_SIZE * 2) {
/* A real frame */
readpos = 0;
+ if (chan->state != AST_STATE_UP) {
+ /* Don't transmit unless it's up */
+ return &f;
+ }
f.frametype = AST_FRAME_VOICE;
f.subclass = AST_FORMAT_SLINEAR;
f.timelen = FRAME_SIZE / 8;
@@ -887,7 +903,7 @@ static int console_dial(int fd, int argc, char *argv[])
if (tmp2 && strlen(tmp2))
myc = tmp2;
}
- if (ast_exists_extension(NULL, myc, mye, 1)) {
+ if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
strncpy(oss.exten, mye, sizeof(oss.exten));
strncpy(oss.context, myc, sizeof(oss.context));
hookstate = 1;
diff --git a/channels/chan_phone.c b/channels/chan_phone.c
index d0ff80843ad0d21938832535236327ee61281cc0..27de1daac78f4c2f88a23aacbe17f7c2f7c37dd8 100755
--- a/channels/chan_phone.c
+++ b/channels/chan_phone.c
@@ -35,6 +35,7 @@
#include "DialTone.h"
#define PHONE_MAX_BUF 480
+#define DEFAULT_GAIN 0x100
static char *desc = "Linux Telephony API Support";
static char *type = "Phone";
@@ -52,7 +53,7 @@ static int echocancel = AEC_OFF;
static int silencesupression = 0;
-static int prefformat = AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR;
+static int prefformat = AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW;
static pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
@@ -90,13 +91,19 @@ static struct phone_pvt {
char buf[PHONE_MAX_BUF]; /* Static buffer for reading frames */
int obuflen;
int dialtone;
+ int txgain, rxgain; /* gain control for playing, recording */
+ /* 0x100 - 1.0, 0x200 - 2.0, 0x80 - 0.5 */
+ int cpt; /* Call Progress Tone playing? */
int silencesupression;
char context[AST_MAX_EXTENSION];
char obuf[PHONE_MAX_BUF * 2];
char ext[AST_MAX_EXTENSION];
char language[MAX_LANGUAGE];
+ char callerid[AST_MAX_EXTENSION];
} *iflist = NULL;
+static char callerid[AST_MAX_EXTENSION];
+
static int phone_digit(struct ast_channel *ast, char digit)
{
struct phone_pvt *p;
@@ -179,6 +186,7 @@ static int phone_hangup(struct ast_channel *ast)
if (option_debug)
ast_log(LOG_DEBUG, "Got hunghup, giving busy signal\n");
ioctl(p->fd, PHONE_BUSY);
+ p->cpt = 1;
}
p->lastformat = -1;
p->lastinput = -1;
@@ -226,6 +234,15 @@ static int phone_setup(struct ast_channel *ast)
return -1;
}
}
+ } else if (ast->pvt->rawreadformat == AST_FORMAT_ULAW) {
+ ioctl(p->fd, PHONE_REC_STOP);
+ if (p->lastinput != AST_FORMAT_ULAW) {
+ p->lastinput = AST_FORMAT_ULAW;
+ if (ioctl(p->fd, PHONE_REC_CODEC, ULAW)) {
+ ast_log(LOG_WARNING, "Failed to set codec to signed linear 16\n");
+ return -1;
+ }
+ }
} else {
ast_log(LOG_WARNING, "Can't do format %d\n", ast->pvt->rawreadformat);
return -1;
@@ -397,8 +414,16 @@ static int phone_write_buf(struct phone_pvt *p, char *buf, int len, int frlen)
#endif
if (res != frlen) {
if (res < 1) {
+/*
+ * Card is in non-blocking mode now and it works well now, but there are
+ * lot of messages like this. So, this message is temporarily disabled.
+ */
+#if 0
ast_log(LOG_WARNING, "Write failed: %s\n", strerror(errno));
return -1;
+#else
+ return 0;
+#endif
} else {
ast_log(LOG_WARNING, "Only wrote %d of %d bytes\n", res, frlen);
}
@@ -427,7 +452,8 @@ static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
ast_frfree(frame);
return -1;
}
- if (!(frame->subclass & (AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR))) {
+ if (!(frame->subclass &
+ (AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW))) {
ast_log(LOG_WARNING, "Cannot handle frames in %d format\n", frame->subclass);
ast_frfree(frame);
return -1;
@@ -479,6 +505,25 @@ static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
p->obuflen = 0;
}
maxfr = 480;
+ } else if (frame->subclass == AST_FORMAT_ULAW) {
+ if (p->lastformat != AST_FORMAT_ULAW) {
+ ioctl(p->fd, PHONE_PLAY_STOP);
+ ioctl(p->fd, PHONE_REC_STOP);
+ if (ioctl(p->fd, PHONE_PLAY_CODEC, ULAW)) {
+ ast_log(LOG_WARNING, "Unable to set uLaw mode\n");
+ return -1;
+ }
+ if (ioctl(p->fd, PHONE_REC_CODEC, ULAW)) {
+ ast_log(LOG_WARNING, "Unable to set uLaw mode\n");
+ return -1;
+ }
+ p->lastformat = AST_FORMAT_ULAW;
+ p->lastinput = AST_FORMAT_ULAW;
+ codecset = 1;
+ /* Reset output buffer */
+ p->obuflen = 0;
+ }
+ maxfr = 240;
}
if (codecset) {
ioctl(p->fd, PHONE_REC_DEPTH, 3);
@@ -517,8 +562,14 @@ static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
if (res != expected) {
if (res < 0)
ast_log(LOG_WARNING, "Write returned error (%s)\n", strerror(errno));
+/*
+ * Card is in non-blocking mode now and it works well now, but there are
+ * lot of messages like this. So, this message is temporarily disabled.
+ */
+#if 0
else
ast_log(LOG_WARNING, "Only wrote %d of %d bytes\n", res, frame->datalen);
+#endif
return -1;
}
sofar += res;
@@ -553,6 +604,8 @@ static struct ast_channel *phone_new(struct phone_pvt *i, int state, char *conte
strncpy(tmp->exten, i->ext, sizeof(tmp->exten));
if (strlen(i->language))
strncpy(tmp->language, i->language, sizeof(tmp->language));
+ if (strlen(i->callerid))
+ tmp->callerid = strdup(i->callerid);
i->owner = tmp;
ast_pthread_mutex_lock(&usecnt_lock);
usecnt++;
@@ -561,6 +614,7 @@ static struct ast_channel *phone_new(struct phone_pvt *i, int state, char *conte
if (state != AST_STATE_DOWN) {
if (state == AST_STATE_RING) {
ioctl(tmp->fds[0], PHONE_RINGBACK);
+ i->cpt = 1;
}
if (ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
@@ -603,7 +657,7 @@ static void phone_check_exception(struct phone_pvt *i)
i->dialtone = 0;
if (strlen(i->ext) < AST_MAX_EXTENSION - 1)
strcat(i->ext, digit);
- if (ast_exists_extension(NULL, i->context, i->ext, 1)) {
+ if (ast_exists_extension(NULL, i->context, i->ext, 1, i->callerid)) {
/* It's a valid extension in its context, get moving! */
phone_new(i, AST_STATE_RING, i->context);
/* No need to restart monitor, we are the monitor */
@@ -613,10 +667,10 @@ static void phone_check_exception(struct phone_pvt *i)
ast_pthread_mutex_unlock(&usecnt_lock);
ast_update_use_count();
}
- } else if (!ast_canmatch_extension(NULL, i->context, i->ext, 1)) {
+ } else if (!ast_canmatch_extension(NULL, i->context, i->ext, 1, i->callerid)) {
/* There is nothing in the specified extension that can match anymore.
Try the default */
- if (ast_exists_extension(NULL, "default", i->ext, 1)) {
+ if (ast_exists_extension(NULL, "default", i->ext, 1, i->callerid)) {
/* Check the default, too... */
phone_new(i, AST_STATE_RING, "default");
if (i->owner) {
@@ -626,11 +680,12 @@ static void phone_check_exception(struct phone_pvt *i)
ast_update_use_count();
}
/* XXX This should probably be justified better XXX */
- } else if (!ast_canmatch_extension(NULL, "default", i->ext, 1)) {
+ } else if (!ast_canmatch_extension(NULL, "default", i->ext, 1, i->callerid)) {
/* It's not a valid extension, give a busy signal */
if (option_debug)
ast_log(LOG_DEBUG, "%s can't match anything in %s or default\n", i->ext, i->context);
ioctl(i->fd, PHONE_BUSY);
+ i->cpt = 1;
}
}
#if 0
@@ -665,7 +720,11 @@ static void phone_check_exception(struct phone_pvt *i)
ast_update_use_count();
}
memset(i->ext, 0, sizeof(i->ext));
- ioctl(i->fd, PHONE_CPT_STOP);
+ if (i->cpt)
+ {
+ ioctl(i->fd, PHONE_CPT_STOP);
+ i->cpt = 0;
+ }
ioctl(i->fd, PHONE_PLAY_STOP);
ioctl(i->fd, PHONE_REC_STOP);
i->dialtone = 0;
@@ -833,13 +892,11 @@ static int restart_monitor()
return 0;
}
-static struct phone_pvt *mkif(char *iface, int mode)
+static struct phone_pvt *mkif(char *iface, int mode, int txgain, int rxgain)
{
/* Make a phone_pvt structure for this interface */
struct phone_pvt *tmp;
-#if 0
int flags;
-#endif
tmp = malloc(sizeof(struct phone_pvt));
if (tmp) {
@@ -852,6 +909,9 @@ static struct phone_pvt *mkif(char *iface, int mode)
if (mode == MODE_FXO) {
if (ioctl(tmp->fd, IXJCTL_PORT, PORT_PSTN))
ast_log(LOG_DEBUG, "Unable to set port to PSTN\n");
+ } else {
+ if (ioctl(tmp->fd, IXJCTL_PORT, PORT_POTS))
+ ast_log(LOG_DEBUG, "Unable to set port to PSTN\n");
}
ioctl(tmp->fd, PHONE_PLAY_STOP);
ioctl(tmp->fd, PHONE_REC_STOP);
@@ -867,10 +927,8 @@ static struct phone_pvt *mkif(char *iface, int mode)
ioctl(tmp->fd, PHONE_VAD, tmp->silencesupression);
#endif
tmp->mode = mode;
-#if 0
flags = fcntl(tmp->fd, F_GETFL);
fcntl(tmp->fd, F_SETFL, flags | O_NONBLOCK);
-#endif
tmp->owner = NULL;
tmp->lastformat = -1;
tmp->lastinput = -1;
@@ -882,6 +940,12 @@ static struct phone_pvt *mkif(char *iface, int mode)
tmp->next = NULL;
tmp->obuflen = 0;
tmp->dialtone = 0;
+ tmp->cpt = 0;
+ strncpy(tmp->callerid, callerid, sizeof(tmp->callerid));
+ tmp->txgain = txgain;
+ ioctl(tmp->fd, PHONE_PLAY_VOLUME, tmp->txgain);
+ tmp->rxgain = rxgain;
+ ioctl(tmp->fd, PHONE_REC_VOLUME, tmp->rxgain);
}
return tmp;
}
@@ -894,7 +958,7 @@ static struct ast_channel *phone_request(char *type, int format, void *data)
char *name = data;
oldformat = format;
- format &= (AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR);
+ format &= (AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW);
if (!format) {
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%d'\n", oldformat);
return NULL;
@@ -919,12 +983,36 @@ static struct ast_channel *phone_request(char *type, int format, void *data)
return tmp;
}
+/* parse gain value from config file */
+static int parse_gain_value(char *gain_type, char *value)
+{
+ float gain;
+
+ /* try to scan number */
+ if (sscanf(value, "%f", &gain) != 1)
+ {
+ ast_log(LOG_ERROR, "Invalid %s value '%s' in '%s' config\n",
+ value, gain_type, config);
+ return DEFAULT_GAIN;
+ }
+
+ /* multiplicate gain by 1.0 gain value */
+ gain = gain * (float)DEFAULT_GAIN;
+
+ /* percentage? */
+ if (value[strlen(value) - 1] == '%')
+ return (int)(gain / (float)100);
+
+ return (int)gain;
+}
+
int load_module()
{
struct ast_config *cfg;
struct ast_variable *v;
struct phone_pvt *tmp;
int mode = MODE_IMMEDIATE;
+ int txgain = DEFAULT_GAIN, rxgain = DEFAULT_GAIN; /* default gain 1.0 */
cfg = ast_load(config);
/* We *must* have a config file otherwise stop immediately */
@@ -941,7 +1029,7 @@ int load_module()
while(v) {
/* Create the interface list */
if (!strcasecmp(v->name, "device")) {
- tmp = mkif(v->value, mode);
+ tmp = mkif(v->value, mode, txgain, rxgain);
if (tmp) {
tmp->next = iflist;
iflist = tmp;
@@ -957,6 +1045,8 @@ int load_module()
silencesupression = ast_true(v->value);
} else if (!strcasecmp(v->name, "language")) {
strncpy(language, v->value, sizeof(language));
+ } else if (!strcasecmp(v->name, "callerid")) {
+ strncpy(callerid, v->value, sizeof(callerid));
} else if (!strcasecmp(v->name, "mode")) {
if (!strncasecmp(v->value, "di", 2))
mode = MODE_DIALTONE;
@@ -988,12 +1078,17 @@ int load_module()
echocancel = AEC_HIGH;
} else
ast_log(LOG_WARNING, "Unknown echo cancellation '%s'\n", v->value);
- }
+ } else if (!strcasecmp(v->name, "txgain")) {
+ txgain = parse_gain_value(v->name, v->value);
+ } else if (!strcasecmp(v->name, "rxgain")) {
+ rxgain = parse_gain_value(v->name, v->value);
+ }
v = v->next;
}
ast_pthread_mutex_unlock(&iflock);
/* Make sure we can register our Adtranphone channel type */
- if (ast_channel_register(type, tdesc, AST_FORMAT_G723_1, phone_request)) {
+ if (ast_channel_register(type, tdesc,
+ AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW, phone_request)) {
ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
ast_destroy(cfg);
unload_module();