diff --git a/cdr/cdr_odbc.c b/cdr/cdr_odbc.c
index 0b601d2925f1786f3b760eff36abf69443844627..7ea2f041febe1b6a7b4e09aab8124a8964684998 100644
--- a/cdr/cdr_odbc.c
+++ b/cdr/cdr_odbc.c
@@ -94,7 +94,7 @@ static SQLHSTMT execute_cb(struct odbc_obj *obj, void *data)
ODBC_res = SQLAllocHandle(SQL_HANDLE_STMT, obj->con, &stmt);
if ((ODBC_res != SQL_SUCCESS) && (ODBC_res != SQL_SUCCESS_WITH_INFO)) {
- ast_verb(11, "cdr_odbc: Failure in AllocStatement %d\n", ODBC_res);
+ ast_log(LOG_WARNING, "cdr_odbc: Failure in AllocStatement %d\n", ODBC_res);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
return NULL;
}
@@ -139,7 +139,7 @@ static SQLHSTMT execute_cb(struct odbc_obj *obj, void *data)
ODBC_res = SQLExecDirect(stmt, (unsigned char *)sqlcmd, SQL_NTS);
if ((ODBC_res != SQL_SUCCESS) && (ODBC_res != SQL_SUCCESS_WITH_INFO)) {
- ast_verb(11, "cdr_odbc: Error in ExecDirect: %d\n", ODBC_res);
+ ast_log(LOG_WARNING, "cdr_odbc: Error in ExecDirect: %d, query is: %s\n", ODBC_res, sqlcmd);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
return NULL;
}
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b372d8c5b40c415cad562f3331792976b7ad2ece..343e66488414c7efc5dfc101284aa3cf7602636c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10032,7 +10032,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
sprintf(offer->decline_m_line, "m=audio 0 %s %s", protocol, codecs);
if (x == 0) {
- ast_log(LOG_WARNING, "Ignoring audio media offer because port number is zero\n");
+ ast_debug(1, "Ignoring audio media offer because port number is zero\n");
continue;
}
@@ -10114,7 +10114,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
sprintf(offer->decline_m_line, "m=video 0 %s %s", protocol, codecs);
if (x == 0) {
- ast_log(LOG_WARNING, "Ignoring video stream offer because port number is zero\n");
+ ast_debug(1, "Ignoring video stream offer because port number is zero\n");
continue;
}
@@ -10192,7 +10192,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
sprintf(offer->decline_m_line, "m=text 0 %s %s", protocol, codecs);
if (x == 0) {
- ast_log(LOG_WARNING, "Ignoring text stream offer because port number is zero\n");
+ ast_debug(1, "Ignoring text stream offer because port number is zero\n");
continue;
}
@@ -10255,7 +10255,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
strcpy(offer->decline_m_line, "m=image 0 udptl t38");
if (x == 0) {
- ast_log(LOG_WARNING, "Ignoring image stream offer because port number is zero\n");
+ ast_debug(1, "Ignoring image stream offer because port number is zero\n");
continue;
}
@@ -10600,7 +10600,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_sockaddr_set_port(isa, udptlportno);
ast_udptl_set_peer(p->udptl, isa);
if (debug)
- ast_debug(1,"Peer T.38 UDPTL is at port %s\n", ast_sockaddr_stringify(isa));
+ ast_debug(1, "Peer T.38 UDPTL is at port %s\n", ast_sockaddr_stringify(isa));
/* verify the far max ifp can be calculated. this requires far max datagram to be set. */
if (!ast_udptl_get_far_max_datagram(p->udptl)) {
@@ -21269,7 +21269,7 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
}
/* Send the feature code to the PBX as DTMF, just like the handset had sent it */
f.len = 100;
- for (j=0; j < strlen(feat->exten); j++) {
+ for (j = 0; j < strlen(feat->exten); j++) {
f.subclass.integer = feat->exten[j];
ast_queue_frame(p->owner, &f);
if (sipdebug) {
@@ -21360,7 +21360,7 @@ static char *sip_do_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args
ast_cli(a->fd, "SIP Debugging Disabled\n");
return CLI_SUCCESS;
}
- } else if (a->argc == e->args +1) {/* ip/peer */
+ } else if (a->argc == e->args + 1) { /* ip/peer */
if (!strcasecmp(what, "ip"))
return sip_do_debug_ip(a->fd, a->argv[e->args]);
else if (!strcasecmp(what, "peer"))
@@ -27559,11 +27559,12 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
accept = __get_header(req, "Accept", &start);
while (!found_supported && !ast_strlen_zero(accept)) {
found_supported = strcmp(accept, "application/simple-message-summary") ? 0 : 1;
- if (!found_supported && (option_debug > 2)) {
- ast_debug(1, "Received SIP mailbox subscription for unknown format: %s\n", accept);
+ if (!found_supported) {
+ ast_debug(3, "Received SIP mailbox subscription for unknown format: %s\n", accept);
}
accept = __get_header(req, "Accept", &start);
}
+ /* If !start, there is no Accept header at all */
if (start && !found_supported) {
/* Format requested that we do not support */
transmit_response(p, "406 Not Acceptable", req);
@@ -32823,7 +32824,7 @@ static void sip_send_all_mwi_subscriptions(void)
static int setup_srtp(struct sip_srtp **srtp)
{
if (!ast_rtp_engine_srtp_is_registered()) {
- ast_log(LOG_ERROR, "No SRTP module loaded, can't setup SRTP session.\n");
+ ast_debug(1, "No SRTP module loaded, can't setup SRTP session.\n");
return -1;
}
diff --git a/main/pbx.c b/main/pbx.c
index 9fdd3a15474b96588dd15155673739f53a5db8b8..1ddda903ca0a22162ee00c46ab684accac3d2eda 100644
--- a/main/pbx.c
+++ b/main/pbx.c
@@ -1580,12 +1580,6 @@ int pbx_exec(struct ast_channel *c, /*!< Channel */
if (app->module)
u = __ast_module_user_add(app->module, c);
- if (strcasecmp(app->name, "system") && !ast_strlen_zero(data) &&
- strchr(data, '|') && !strchr(data, ',') && !ast_opt_dont_warn) {
- ast_log(LOG_WARNING, "The application delimiter is now the comma, not "
- "the pipe. Did you forget to convert your dialplan? (%s(%s))\n",
- app->name, (char *) data);
- }
res = app->execute(c, S_OR(data, ""));
if (app->module && u)
__ast_module_user_remove(app->module, u);
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 00f909fae481771935421590f2decf622fa61ec9..45df0f92382447e73c65fb3039fcdc5685f2a15d 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -3467,7 +3467,14 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
/* Make sure the data that was read in is actually enough to make up an RTP packet */
if (res < hdrlen) {
- ast_log(LOG_WARNING, "RTP Read too short\n");
+ /* If this is a keepalive containing only nulls, don't bother with a warning */
+ int i;
+ for (i = 0; i < res; ++i) {
+ if (rtp->rawdata[AST_FRIENDLY_OFFSET + i] != '\0') {
+ ast_log(LOG_WARNING, "RTP Read too short\n");
+ return &ast_null_frame;
+ }
+ }
return &ast_null_frame;
}