--- asterisk-11.0.1/apps/app_jack.c	2012-03-24 04:03:20.000000000 +0100
+++ asterisk-11.0.1-widebandjack/apps/app_jack.c	2012-11-18 17:06:03.053879363 +0100
@@ -5,6 +5,9 @@
  *
  * Russell Bryant <russell@digium.com>
  *
+ * Variable Bandwidth Modification by
+ * Mathis Schmieder <mathis.schmieder@gmail.com>
+ *
  * See http://www.asterisk.org for more information about
  * the Asterisk project. Please do not directly contact
  * any of the maintainers of this project for assistance;
@@ -73,7 +76,9 @@
 "    n         - Do not automatically start the JACK server if it is not already\n" \
 "                running.\n" \
 "    c(<name>) - By default, Asterisk will use the channel name for the jack client\n" \
-"                name.  Use this option to specify a custom client name.\n"
+"                name.  Use this option to specify a custom client name.\n" \
+"    r(<rate>) - By default, the audiohook will be narrow band. Specify a\n" \
+"                sample rate in Hz to change this.\n"
 /*** DOCUMENTATION
 	<application name="JACK" language="en_US">
 		<synopsis>
@@ -103,6 +108,11 @@
 							<para>Use this option to specify a custom client name.</para>
 						</argument>
 					</option>
+          <option name="r">
+            <argument name="rate" required="true">
+              <para>By default, the audiohook will be narrow band. Specify a sample rate in Hz to change this.</para>
+            </argument>
+          </option>
 				</optionlist>
 			</parameter>
 		</syntax>
@@ -115,6 +125,8 @@
  ***/
 
 static const char jack_app[] = "JACK";
+static double sample_rate = 8000.00;
+static enum ast_format_id channel_format = AST_FORMAT_SLINEAR;
 
 struct jack_data {
 	AST_DECLARE_STRING_FIELDS(
@@ -201,10 +213,8 @@
 
 	jack_srate = jack_get_sample_rate(jack_data->client);
 
-	/* XXX Hard coded 8 kHz */
-
-	to_srate = input ? 8000.0 : jack_srate;
-	from_srate = input ? jack_srate : 8000.0;
+	to_srate = input ? sample_rate : jack_srate;
+	from_srate = input ? jack_srate : sample_rate;
 
 	resample_factor = input ? &jack_data->input_resample_factor :
 		&jack_data->output_resample_factor;
@@ -610,7 +620,7 @@
 		.datalen = sizeof(buf),
 		.samples = ARRAY_LEN(buf),
 	};
-	ast_format_set(&f.subclass.format, AST_FORMAT_SLINEAR, 0);
+	ast_format_set(&f.subclass.format, channel_format, 0);
 
 	for (;;) {
 		size_t res, read_len;
@@ -653,6 +663,7 @@
 	OPT_OUTPUT_PORT =    (1 << 2),
 	OPT_NOSTART_SERVER = (1 << 3),
 	OPT_CLIENT_NAME =    (1 << 4),
+  OPT_SAMPLE_RATE =   (1 << 5),
 };
 
 enum {
@@ -660,6 +671,7 @@
 	OPT_ARG_INPUT_PORT,
 	OPT_ARG_OUTPUT_PORT,
 	OPT_ARG_CLIENT_NAME,
+  OPT_ARG_SAMPLE_RATE,
 
 	/* Must be the last element */
 	OPT_ARG_ARRAY_SIZE,
@@ -671,6 +683,7 @@
 	AST_APP_OPTION_ARG('o', OPT_OUTPUT_PORT, OPT_ARG_OUTPUT_PORT),
 	AST_APP_OPTION('n', OPT_NOSTART_SERVER),
 	AST_APP_OPTION_ARG('c', OPT_CLIENT_NAME, OPT_ARG_CLIENT_NAME),
+  AST_APP_OPTION_ARG('r', OPT_SAMPLE_RATE, OPT_ARG_SAMPLE_RATE),
 END_OPTIONS );
 
 static struct jack_data *jack_data_alloc(void)
@@ -733,6 +746,17 @@
 		}
 	}
 
+	if (ast_test_flag(&options, OPT_SAMPLE_RATE)) {
+		if (!ast_strlen_zero(option_args[OPT_ARG_SAMPLE_RATE])) {
+      sample_rate = atof(option_args[OPT_ARG_SAMPLE_RATE]);
+      channel_format = ast_format_slin_by_rate(sample_rate);
+    }
+		else {
+			ast_log(LOG_ERROR, "A sample rate must be provided with the r() option\n");
+			return -1;
+		}
+	}
+
 	jack_data->no_start_server = ast_test_flag(&options, OPT_NOSTART_SERVER) ? 1 : 0;
 
 	return 0;
@@ -755,12 +779,12 @@
 		return -1;
 	}
 
-	if (ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR)) {
+	if (ast_set_read_format_by_id(chan, channel_format)) {
 		destroy_jack_data(jack_data);
 		return -1;
 	}
 
-	if (ast_set_write_format_by_id(chan, AST_FORMAT_SLINEAR)) {
+	if (ast_set_write_format_by_id(chan, channel_format)) {
 		destroy_jack_data(jack_data);
 		return -1;
 	}
@@ -824,8 +848,9 @@
 	if (frame->frametype != AST_FRAME_VOICE)
 		return 0;
 
-	if (frame->subclass.format.id != AST_FORMAT_SLINEAR) {
-		ast_log(LOG_WARNING, "Expected frame in SLINEAR for the audiohook, but got format %s\n",
+  /* TODO get name of wanted format */
+	if (frame->subclass.format.id != channel_format) {
+		ast_log(LOG_WARNING, "Audiohook format mismatch. Got unexpected format %s\n",
 			ast_getformatname(&frame->subclass.format));
 		return 0;
 	}