Change a few warnings to debug and the inverse.

Remove the "RTP Read too short" warning for RTP keepalives. Remove the
the warning about the application delimiter switch from pipe to comma.
(You should've done this by now.) Make cdr_odbc report more when an
insert fails. Make chan_sip warn less when the peer wants SRTP (and we
don't) or sends a zero port to disable a media type.

Review: https://reviewboard.asterisk.org/r/2167
(closes issue ASTERISK-20538)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375137 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/cdr/cdr_odbc.c b/cdr/cdr_odbc.c
index 0b601d2..7ea2f04 100644 (file)
--- 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 b372d8c..343e664 100644 (file)
--- 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 9fdd3a1..1ddda90 100644 (file)
--- 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 00f909f..45df0f9 100644 (file)
--- 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;
        }