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Added more logging for Asterisk AMD as well as beta AMD patch for logging of NOAUDIODATA answered calls and more AMD stats gathering

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Added new AST_activity_log_export.pl script to export selected data and send to an FTP server

git-svn-id: svn://192.168.202.10@3082 3d104415-ff17-0410-8863-d5cf3c621b8a
This commit is contained in:
mattf
2019-03-27 16:49:15 +00:00
parent 5d08d54796
commit e90ac24b7d
7 changed files with 981 additions and 8 deletions
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extras/ast13_amd_noaudio.patch
+78
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@@ -0,0 +1,78 @@
--- asterisk-13.21.1/apps/app_amd.c 2018-06-11 17:02:31.000000000 -0400
+++ asterisk-13.21.1.new/apps/app_amd.c 2019-03-20 13:02:25.675807594 -0400
@@ -45,6 +45,7 @@
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+#include <sys/time.h>
#include "asterisk/module.h"
#include "asterisk/lock.h"
#include "asterisk/channel.h"
@@ -164,6 +165,7 @@
static void isAnsweringMachine(struct ast_channel *chan, const char *data)
{
int res = 0;
+ int audioFrameCount = 0;
struct ast_frame *f = NULL;
struct ast_dsp *silenceDetector = NULL;
int dspsilence = 0, framelength = 0;
@@ -176,7 +178,7 @@
int iWordsCount = 0;
int currentState = STATE_IN_WORD;
int consecutiveVoiceDuration = 0;
- char amdCause[256] = "", amdStatus[256] = "";
+ char amdCause[256] = "", amdStatus[256] = "", amdStats[256] = "";
char *parse = ast_strdupa(data);
/* Lets set the initial values of the variables that will control the algorithm.
@@ -277,6 +279,9 @@
/* Set silence threshold to specified value */
ast_dsp_set_threshold(silenceDetector, silenceThreshold);
+ /* Set our start time so we can tie the loop to real world time and not RTP updates */
+ struct timeval amd_tvstart = ast_tvnow();
+
/* Now we go into a loop waiting for frames from the channel */
while ((res = ast_waitfor(chan, 2 * maxWaitTimeForFrame)) > -1) {
int ms = 0;
@@ -295,7 +300,24 @@
break;
}
+ /* Check to make sure we haven't gone over our real-world timeout in case frames get stalled for whatever reason */
+ if ( (ast_tvdiff_ms(ast_tvnow(), amd_tvstart)) > totalAnalysisTime ) {
+ ast_frfree(f);
+ strcpy(amdStatus , "NOTSURE");
+ if ( audioFrameCount == 0 ) {
+ ast_verb(3, "AMD: Channel [%s]. No audio date recieved in [%d] seconds.\n", ast_channel_name(chan), totalAnalysisTime);
+ sprintf(amdCause , "NOAUDIODATA-%d", iTotalTime);
+ break;
+ }
+ ast_verb(3, "AMD: Channel [%s]. Timeout...\n", ast_channel_name(chan));
+ sprintf(amdCause , "TOOLONG-%d", iTotalTime);
+ break;
+ }
+
if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_CNG) {
+ /* keep track of the number of audio frames we get */
+ audioFrameCount++;
+
/* Figure out how long the frame is in milliseconds */
if (f->frametype == AST_FRAME_VOICE) {
framelength = (ast_codec_samples_count(f) / DEFAULT_SAMPLES_PER_MS);
@@ -423,9 +445,15 @@
sprintf(amdCause , "TOOLONG-%d", iTotalTime);
}
+ sprintf( amdStats, "%d-%d-%d-%d-%d-%d-%d-%d-%d",
+ iTotalTime, silenceDuration, initialSilence,
+ afterGreetingSilence, voiceDuration, consecutiveVoiceDuration,
+ greeting, iWordsCount, maximumNumberOfWords);
+
/* Set the status and cause on the channel */
pbx_builtin_setvar_helper(chan , "AMDSTATUS" , amdStatus);
pbx_builtin_setvar_helper(chan , "AMDCAUSE" , amdCause);
+ pbx_builtin_setvar_helper(chan , "AMDSTATS" , amdStats);
/* Restore channel read format */
if (readFormat && ast_set_read_format(chan, readFormat))