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-rw-r--r--main/rtp.c25
-rw-r--r--main/say.c4
-rw-r--r--main/sched.c30
3 files changed, 28 insertions, 31 deletions
diff --git a/main/rtp.c b/main/rtp.c
index 04f8e9629..691d39160 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -730,7 +730,7 @@ static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *
event = data[3] & 0x1f;
if (option_debug > 2 || rtpdebug)
- ast_log(LOG_DEBUG, "Cisco DTMF Digit: %02x (len=%d, seq=%d, flags=%02x, power=%d, history count=%d)\n", event, len, seq, flags, power, (len - 4) / 2);
+ ast_debug(0, "Cisco DTMF Digit: %02x (len=%d, seq=%d, flags=%02x, power=%d, history count=%d)\n", event, len, seq, flags, power, (len - 4) / 2);
if (event < 10) {
resp = '0' + event;
} else if (event < 11) {
@@ -790,7 +790,7 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
/* Print out debug if turned on */
if (rtpdebug || option_debug > 2)
- ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
+ ast_debug(0, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
/* Figure out what digit was pressed */
if (event < 10) {
@@ -843,7 +843,7 @@ static struct ast_frame *process_rfc3389(struct ast_rtp *rtp, unsigned char *dat
totally help us out becuase we don't have an engine to keep it going and we are not
guaranteed to have it every 20ms or anything */
if (rtpdebug)
- ast_log(LOG_DEBUG, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", rtp->lastrxformat, len);
+ ast_debug(0, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", rtp->lastrxformat, len);
if (!(ast_test_flag(rtp, FLAG_3389_WARNING))) {
ast_log(LOG_NOTICE, "Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: %s\n",
@@ -933,7 +933,7 @@ struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp)
(rtp->rtcp->them.sin_port != sin.sin_port)) {
memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
if (option_debug || rtpdebug)
- ast_log(LOG_DEBUG, "RTCP NAT: Got RTCP from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
+ ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
}
}
@@ -1198,7 +1198,7 @@ static int bridge_p2p_rtp_write(struct ast_rtp *rtp, struct ast_rtp *bridged, un
ast_debug(1, "RTP Transmission error of packet to %s:%d: %s\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port), strerror(errno));
} else if (((ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(bridged, FLAG_NAT_INACTIVE_NOWARN)) {
if (option_debug || rtpdebug)
- ast_log(LOG_DEBUG, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port));
+ ast_debug(0, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port));
ast_set_flag(bridged, FLAG_NAT_INACTIVE_NOWARN);
}
return 0;
@@ -1284,7 +1284,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
rtp->rxseqno = 0;
ast_set_flag(rtp, FLAG_NAT_ACTIVE);
if (option_debug || rtpdebug)
- ast_log(LOG_DEBUG, "RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
+ ast_debug(0, "RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
}
}
@@ -1306,7 +1306,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
if (option_debug || rtpdebug)
- ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
+ ast_debug(0, "Forcing Marker bit, because SSRC has changed\n");
mark = 1;
}
@@ -1330,9 +1330,9 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
int profile;
profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16;
if (profile == 0x505a)
- ast_log(LOG_DEBUG, "Found Zfone extension in RTP stream - zrtp - not supported.\n");
+ ast_debug(1, "Found Zfone extension in RTP stream - zrtp - not supported.\n");
else
- ast_log(LOG_DEBUG, "Found unknown RTP Extensions %x\n", profile);
+ ast_debug(1, "Found unknown RTP Extensions %x\n", profile);
}
}
@@ -2833,7 +2833,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
} else if (((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(rtp, FLAG_NAT_INACTIVE_NOWARN)) {
/* Only give this error message once if we are not RTP debugging */
if (option_debug || rtpdebug)
- ast_log(LOG_DEBUG, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
+ ast_debug(0, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
}
} else {
@@ -3204,7 +3204,7 @@ static int p2p_rtp_callback(int *id, int fd, short events, void *cbdata)
rtp->rxseqno = 0;
ast_set_flag(rtp, FLAG_NAT_ACTIVE);
if (option_debug || rtpdebug)
- ast_log(LOG_DEBUG, "P2P RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
+ ast_debug(0, "P2P RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
}
/* Write directly out to other RTP stream if bridged */
@@ -3566,8 +3566,7 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
fmt0 = ast_codec_pref_getsize(&p0->pref, c0->rawreadformat);
fmt1 = ast_codec_pref_getsize(&p1->pref, c1->rawreadformat);
if (fmt0.cur_ms != fmt1.cur_ms) {
- if (option_debug)
- ast_log(LOG_DEBUG, "Cannot packet2packet bridge - packetization settings prevent it\n");
+ ast_debug(1, "Cannot packet2packet bridge - packetization settings prevent it\n");
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
diff --git a/main/say.c b/main/say.c
index 7dce69a5c..bbc7239d5 100644
--- a/main/say.c
+++ b/main/say.c
@@ -1381,7 +1381,7 @@ static int ast_say_number_full_hu(struct ast_channel *chan, int num, const char
num = num % 1000000;
snprintf(fn, sizeof(fn), "digits/million");
} else {
- ast_log(LOG_DEBUG, "Number '%d' is too big for me\n", num);
+ ast_debug(1, "Number '%d' is too big for me\n", num);
res = -1;
}
}
@@ -6305,7 +6305,7 @@ static int gr_say_number_female(int num, struct ast_channel *chan, const char *i
int res;
char fn[256] = "";
- /* ast_log(LOG_DEBUG, "\n\n Saying number female %s %d \n\n",lang, num); */
+ /* ast_debug(1, "\n\n Saying number female %s %d \n\n",lang, num); */
if (num < 5) {
snprintf(fn, sizeof(fn), "digits/female-%d", num);
res = wait_file(chan, ints, fn, lang);
diff --git a/main/sched.c b/main/sched.c
index 1dd73334f..1a07ab659 100644
--- a/main/sched.c
+++ b/main/sched.c
@@ -196,7 +196,7 @@ static int sched_settime(struct timeval *tv, int when)
{
struct timeval now = ast_tvnow();
- /*ast_log(LOG_DEBUG, "TV -> %lu,%lu\n", tv->tv_sec, tv->tv_usec);*/
+ /*ast_debug(1, "TV -> %lu,%lu\n", tv->tv_sec, tv->tv_usec);*/
if (ast_tvzero(*tv)) /* not supplied, default to now */
*tv = now;
*tv = ast_tvadd(*tv, ast_samp2tv(when, 1000));
@@ -301,22 +301,20 @@ void ast_sched_dump(const struct sched_context *con)
ast_debug(1, "Asterisk Schedule Dump (%d in Q, %d Total)\n", con->schedcnt, con->eventcnt - 1);
#endif
- if (option_debug) {
- ast_log(LOG_DEBUG, "=============================================================\n");
- ast_log(LOG_DEBUG, "|ID Callback Data Time (sec:ms) |\n");
- ast_log(LOG_DEBUG, "+-----+-----------------+-----------------+-----------------+\n");
- AST_LIST_TRAVERSE(&con->schedq, q, list) {
- struct timeval delta = ast_tvsub(q->when, tv);
-
- ast_log(LOG_DEBUG, "|%.4d | %-15p | %-15p | %.6ld : %.6ld |\n",
- q->id,
- q->callback,
- q->data,
- delta.tv_sec,
- (long int)delta.tv_usec);
- }
- ast_log(LOG_DEBUG, "=============================================================\n");
+ ast_debug(1, "=============================================================\n");
+ ast_debug(1, "|ID Callback Data Time (sec:ms) |\n");
+ ast_debug(1, "+-----+-----------------+-----------------+-----------------+\n");
+ AST_LIST_TRAVERSE(&con->schedq, q, list) {
+ struct timeval delta = ast_tvsub(q->when, tv);
+
+ ast_debug(1, "|%.4d | %-15p | %-15p | %.6ld : %.6ld |\n",
+ q->id,
+ q->callback,
+ q->data,
+ delta.tv_sec,
+ (long int)delta.tv_usec);
}
+ ast_debug(1, "=============================================================\n");
}
/*! \brief
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-29 06:22:23 +0000