mattf
036cbb1fa5
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
79 lines
3.0 KiB
Diff
79 lines
3.0 KiB
Diff
--- asterisk-13.21.1/apps/app_amd.c 2018-06-11 17:02:31.000000000 -0400
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+++ asterisk-13.21.1.new/apps/app_amd.c 2019-03-20 13:02:25.675807594 -0400
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@@ -45,6 +45,7 @@
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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+#include <sys/time.h>
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#include "asterisk/module.h"
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#include "asterisk/lock.h"
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#include "asterisk/channel.h"
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@@ -164,6 +165,7 @@
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static void isAnsweringMachine(struct ast_channel *chan, const char *data)
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{
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int res = 0;
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+ int audioFrameCount = 0;
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struct ast_frame *f = NULL;
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struct ast_dsp *silenceDetector = NULL;
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int dspsilence = 0, framelength = 0;
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@@ -176,7 +178,7 @@
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int iWordsCount = 0;
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int currentState = STATE_IN_WORD;
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int consecutiveVoiceDuration = 0;
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- char amdCause[256] = "", amdStatus[256] = "";
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+ char amdCause[256] = "", amdStatus[256] = "", amdStats[256] = "";
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char *parse = ast_strdupa(data);
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/* Lets set the initial values of the variables that will control the algorithm.
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@@ -277,6 +279,9 @@
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/* Set silence threshold to specified value */
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ast_dsp_set_threshold(silenceDetector, silenceThreshold);
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+ /* Set our start time so we can tie the loop to real world time and not RTP updates */
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+ struct timeval amd_tvstart = ast_tvnow();
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+
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/* Now we go into a loop waiting for frames from the channel */
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while ((res = ast_waitfor(chan, 2 * maxWaitTimeForFrame)) > -1) {
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int ms = 0;
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@@ -295,7 +300,24 @@
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break;
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}
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+ /* Check to make sure we haven't gone over our real-world timeout in case frames get stalled for whatever reason */
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+ if ( (ast_tvdiff_ms(ast_tvnow(), amd_tvstart)) > totalAnalysisTime ) {
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+ ast_frfree(f);
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+ strcpy(amdStatus , "NOTSURE");
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+ if ( audioFrameCount == 0 ) {
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+ ast_verb(3, "AMD: Channel [%s]. No audio date recieved in [%d] seconds.\n", ast_channel_name(chan), totalAnalysisTime);
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+ sprintf(amdCause , "NOAUDIODATA-%d", iTotalTime);
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+ break;
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+ }
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+ ast_verb(3, "AMD: Channel [%s]. Timeout...\n", ast_channel_name(chan));
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+ sprintf(amdCause , "TOOLONG-%d", iTotalTime);
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+ break;
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+ }
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+
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if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_CNG) {
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+ /* keep track of the number of audio frames we get */
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+ audioFrameCount++;
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+
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/* Figure out how long the frame is in milliseconds */
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if (f->frametype == AST_FRAME_VOICE) {
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framelength = (ast_codec_samples_count(f) / DEFAULT_SAMPLES_PER_MS);
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@@ -423,9 +445,15 @@
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sprintf(amdCause , "TOOLONG-%d", iTotalTime);
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}
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+ sprintf( amdStats, "%d-%d-%d-%d-%d-%d-%d-%d-%d",
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+ iTotalTime, silenceDuration, initialSilence,
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+ afterGreetingSilence, voiceDuration, consecutiveVoiceDuration,
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+ greeting, iWordsCount, maximumNumberOfWords);
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+
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/* Set the status and cause on the channel */
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pbx_builtin_setvar_helper(chan , "AMDSTATUS" , amdStatus);
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pbx_builtin_setvar_helper(chan , "AMDCAUSE" , amdCause);
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+ pbx_builtin_setvar_helper(chan , "AMDSTATS" , amdStats);
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/* Restore channel read format */
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if (readFormat && ast_set_read_format(chan, readFormat))
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