Asterisk
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asterisk
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Show uptime

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.0
Mark Spencer 22 years ago
parent 163898dae7
commit fba6a02260
6 changed files with 130 additions and 1 deletions
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4
asterisk.c
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@ -70,6 +70,9 @@ struct console {
pthread_t t; /* Thread of handler */
};
time_t ast_startuptime;
time_t ast_lastreloadtime;
static History *el_hist = NULL;
static EditLine *el = NULL;
static char *remotehostname;
@ -1349,6 +1352,7 @@ int main(int argc, char *argv[])
#ifdef __AST_DEBUG_MALLOC
__ast_mm_init();
#endif
time(&ast_startuptime);
ast_cli_register(&astshutdownnow);
ast_cli_register(&astshutdowngracefully);
ast_cli_register(&astrestartnow);

87
cli.c
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@ -173,6 +173,92 @@ static char version_help[] =
"Usage: show version\n"
" Shows Asterisk version information.\n ";
static char *format_uptimestr(time_t timeval)
{
int years = 0, weeks = 0, days = 0, hours = 0, mins = 0, secs = 0;
char timestr[256];
int pos = 0;
#define SECOND (1)
#define MIN (SECOND*60)
#define HOUR (MIN*60)
#define DAY (HOUR*24)
#define WEEK (DAY*7)
#define YEAR (DAY*365)
if (timeval > YEAR) {
years = (timeval / YEAR);
timeval -= (years * YEAR);
if (years > 1)
pos += sprintf(timestr + pos, "%d years, ", years);
else
pos += sprintf(timestr + pos, "1 year, ");
}
if (timeval > WEEK) {
weeks = (timeval / WEEK);
timeval -= (weeks * WEEK);
if (weeks > 1)
pos += sprintf(timestr + pos, "%d weeks, ", weeks);
else
pos += sprintf(timestr + pos, "1 week, ");
}
if (timeval > DAY) {
days = (timeval / DAY);
timeval -= (days * DAY);
if (days > 1)
pos += sprintf(timestr + pos, "%d days, ", days);
else
pos += sprintf(timestr + pos, "1 day, ");
}
if (timeval > HOUR) {
hours = (timeval / HOUR);
timeval -= (hours * HOUR);
if (hours > 1)
pos += sprintf(timestr + pos, "%d hours, ", hours);
else
pos += sprintf(timestr + pos, "1 hour, ");
}
if (timeval > MIN) {
mins = (timeval / MIN);
timeval -= (mins * MIN);
if (mins > 1)
pos += sprintf(timestr + pos, "%d minutes, ", mins);
else if (mins > 0)
pos += sprintf(timestr + pos, "1 minute, ");
}
secs = timeval;
if (secs > 0)
pos += sprintf(timestr + pos, "%d seconds", secs);
return timestr ? strdup(timestr) : NULL;
}
static int handle_showuptime(int fd, int argc, char *argv[])
{
time_t curtime, tmptime;
char *timestr;
time(&curtime);
if (ast_startuptime) {
tmptime = curtime - ast_startuptime;
timestr = format_uptimestr(tmptime);
if (timestr) {
ast_cli(fd, "System uptime: %s\n", timestr);
free(timestr);
}
}
if (ast_lastreloadtime) {
tmptime = curtime - ast_lastreloadtime;
timestr = format_uptimestr(tmptime);
if (timestr) {
ast_cli(fd, "Last reload: %s\n", timestr);
free(timestr);
}
}
return RESULT_SUCCESS;
}
static int handle_modlist(int fd, int argc, char *argv[])
{
if (argc != 2)
@ -476,6 +562,7 @@ static struct ast_cli_entry builtins[] = {
{ { "show", "channel", NULL }, handle_showchan, "Display information on a specific channel", showchan_help, complete_ch },
{ { "show", "channels", NULL }, handle_chanlist, "Display information on channels", chanlist_help },
{ { "show", "modules", NULL }, handle_modlist, "List modules and info", modlist_help },
{ { "show", "uptime", NULL }, handle_showuptime, "Show uptime information", modlist_help },
{ { "show", "version", NULL }, handle_version, "Display version info", version_help },
{ { "soft", "hangup", NULL }, handle_softhangup, "Request a hangup on a given channel", softhangup_help, complete_ch },
{ { "unload", NULL }, handle_unload, "Unload a dynamic module by name", unload_help, complete_fn },

2
include/asterisk/options.h
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@ -27,6 +27,8 @@ extern int option_initcrypto;
extern int option_nocolor;
extern int fully_booted;
extern char defaultlanguage[];
extern time_t ast_startuptime;
extern time_t ast_lastreloadtime;
#define VERBOSE_PREFIX_1 " "
#define VERBOSE_PREFIX_2 " == "

4
include/asterisk/rtp.h
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@ -30,8 +30,10 @@ extern "C" {
#define AST_RTP_DTMF (1 << 0)
/*! 'Comfort Noise' (RFC3389) */
#define AST_RTP_CN (1 << 1)
/*! DTMF (Cisco Proprietary) */
#define AST_RTP_CISCO_DTMF (1 << 2)
/*! Maximum RTP-specific code */
#define AST_RTP_MAX AST_RTP_CN
#define AST_RTP_MAX AST_RTP_CISCO_DTMF
struct ast_rtp_protocol {
struct ast_rtp *(*get_rtp_info)(struct ast_channel *chan); /* Get RTP struct, or NULL if unwilling to transfer */

1
loader.c
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@ -146,6 +146,7 @@ void ast_module_reload(void)
/* We'll do the logger and manager the favor of calling its reload here first */
reload_manager();
ast_enum_reload();
time(&ast_lastreloadtime);
ast_pthread_mutex_lock(&modlock);
m = module_list;

33
rtp.c
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@ -151,6 +151,33 @@ static struct ast_frame *send_dtmf(struct ast_rtp *rtp)
}
static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *data, int len)
{
unsigned int event;
char resp = 0;
struct ast_frame *f = NULL;
event = ntohl(*((unsigned int *)(data)));
event &= 0x001F;
#if 0
printf("Cisco Digit: %08x (len = %d)\n", event, len);
#endif
if (event < 10) {
resp = '0' + event;
} else if (event < 11) {
resp = '*';
} else if (event < 12) {
resp = '#';
} else if (event < 16) {
resp = 'A' + (event - 12);
}
if (rtp->resp && (rtp->resp != resp)) {
f = send_dtmf(rtp);
}
rtp->resp = resp;
rtp->dtmfcount = dtmftimeout;
return f;
}
static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len)
{
unsigned int event;
@ -287,6 +314,10 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
/* It's special -- rfc2833 process it */
f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
if (f) return f; else return &null_frame;
} else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
/* It's really special -- process it the Cisco way */
f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
if (f) return f; else return &null_frame;
} else if (rtpPT.code == AST_RTP_CN) {
/* Comfort Noise */
f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
@ -379,6 +410,7 @@ static struct {
{{1, AST_FORMAT_SPEEX}, "audio", "SPEEX"},
{{1, AST_FORMAT_ILBC}, "audio", "iLBC"},
{{0, AST_RTP_DTMF}, "audio", "telephone-event"},
{{0, AST_RTP_CISCO_DTMF}, "audio", "bastard-telephone-event"},
{{0, AST_RTP_CN}, "audio", "CN"},
{{1, AST_FORMAT_JPEG}, "video", "JPEG"},
{{1, AST_FORMAT_PNG}, "video", "PNG"},
@ -410,6 +442,7 @@ static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
[97] = {1, AST_FORMAT_ILBC},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},
[121] = {0, AST_RTP_CISCO_DTMF}, // Must be type 121
};
void ast_rtp_pt_clear(struct ast_rtp* rtp)

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