merge rev 44253 - option_debug checking before printing to debug channel.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44661 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f75a71a73ce675fd20e8a6ca34a5ae0b7f3f0968..3ca2e0a8020adbf70ec04baa0b67e5ae84ffc58a 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1574,7 +1574,7 @@ static char *referstatus2str(enum referstatus rstatus)
        a dialog */
 static void initialize_initreq(struct sip_pvt *p, struct sip_request *req)
 {
-       if (p->initreq.headers) {
+       if (p->initreq.headers && option_debug) {
                ast_log(LOG_DEBUG, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
        }
        /* Use this as the basis */
@@ -2007,6 +2007,7 @@ static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod, int
                        ((ast_test_flag(cur, FLAG_RESPONSE)) || 
                         (!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
                        if (!resp && (seqno == p->pendinginvite)) {
+                               if (option_debug)
                                ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
                                p->pendinginvite = 0;
                        }
@@ -3174,6 +3175,7 @@ static const char *hangup_cause2sip(int cause)
                        
                case AST_CAUSE_NOTDEFINED:
                default:
+                       if (option_debug)
                        ast_log(LOG_DEBUG, "AST hangup cause %d (no match found in SIP)\n", cause);
                        return NULL;
        }
@@ -3193,6 +3195,7 @@ static int sip_hangup(struct ast_channel *ast)
        struct ast_channel *oldowner = ast;
 
        if (!p) {
+               if (option_debug)
                ast_log(LOG_DEBUG, "Asked to hangup channel that was not connected\n");
                return 0;
        }
@@ -3212,9 +3215,11 @@ static int sip_hangup(struct ast_channel *ast)
        if (option_debug) {
                if (ast_test_flag(ast, AST_FLAG_ZOMBIE) && p->refer && option_debug)
                        ast_log(LOG_DEBUG, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid);
-               else 
+               else  {
+                       if (option_debug)
                        ast_log(LOG_DEBUG, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid);
        }
+       }
        if (option_debug && ast_test_flag(ast, AST_FLAG_ZOMBIE)) 
                ast_log(LOG_DEBUG, "Hanging up zombie call. Be scared.\n");
 
@@ -3459,9 +3464,9 @@ static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
        int ret = -1;
        struct sip_pvt *p;
 
-       if (newchan && ast_test_flag(newchan, AST_FLAG_ZOMBIE))
+       if (newchan && ast_test_flag(newchan, AST_FLAG_ZOMBIE) && option_debug)
                ast_log(LOG_DEBUG, "New channel is zombie\n");
-       if (oldchan && ast_test_flag(oldchan, AST_FLAG_ZOMBIE))
+       if (oldchan && ast_test_flag(oldchan, AST_FLAG_ZOMBIE) && option_debug)
                ast_log(LOG_DEBUG, "Old channel is zombie\n");
 
        if (!newchan || !newchan->tech_pvt) {
@@ -4793,6 +4798,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                                framing = strtol(tmp, NULL, 10);
                                if (framing == LONG_MIN || framing == LONG_MAX) {
                                        framing = 0;
+                                       if (option_debug)
                                        ast_log(LOG_DEBUG, "Can't read framing from SDP: %s\n", a);
                                }
                        }
@@ -7040,6 +7046,7 @@ static int transmit_register(struct sip_registry *r, int sipmethod, const char *
                        ast_sched_del(sched, r->timeout);
                }
                r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, r);
+               if (option_debug)
                ast_log(LOG_DEBUG, "Scheduled a registration timeout for %s id  #%d \n", r->hostname, r->timeout);
        }
 
@@ -7649,6 +7656,7 @@ static void build_route(struct sip_pvt *p, struct sip_request *req, int backward
 
        /* Once a persistant route is set, don't fool with it */
        if (p->route && p->route_persistant) {
+               if (option_debug)
                ast_log(LOG_DEBUG, "build_route: Retaining previous route: <%s>\n", p->route->hop);
                return;
        }
@@ -8246,6 +8254,7 @@ static int get_destination(struct sip_pvt *p, struct sip_request *oreq)
                domain_context[0] = '\0';
                if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) {
                        if (!allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) {
+                               if (option_debug)
                                ast_log(LOG_DEBUG, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain);
                                return -2;
                        }
@@ -8899,14 +8908,17 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
                        }
                        usenatroute = ast_test_flag(&p->flags[0], SIP_NAT_ROUTE);
                        if (p->rtp) {
+                               if (option_debug)
                                ast_log(LOG_DEBUG, "Setting NAT on RTP to %s\n", usenatroute ? "On" : "Off");
                                ast_rtp_setnat(p->rtp, usenatroute);
                        }
                        if (p->vrtp) {
+                               if (option_debug)
                                ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", usenatroute ? "On" : "Off");
                                ast_rtp_setnat(p->vrtp, usenatroute);
                        }
                        if (p->udptl) {
+                               if (option_debug)
                                ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", usenatroute ? "On" : "Off");
                                ast_udptl_setnat(p->udptl, usenatroute);
                        }
@@ -10441,6 +10453,11 @@ static void sip_dump_history(struct sip_pvt *dialog)
        if (!dialog)
                return;
 
+       if (!option_debug && !sipdebug) {
+               ast_log(LOG_NOTICE, "You must have debugging enabled (SIP or Asterisk) in order to dump SIP history.\n");
+               return;
+       }
+
        ast_log(LOG_DEBUG, "\n---------- SIP HISTORY for '%s' \n", dialog->callid);
        if (dialog->subscribed)
                ast_log(LOG_DEBUG, "  * Subscription\n");
@@ -11271,6 +11288,7 @@ static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req)
                e = strchr(s, '/');
                if (e)
                        *e = '\0';
+               if (option_debug)
                ast_log(LOG_DEBUG, "Found promiscuous redirection to 'SIP/%s'\n", s);
                if (p->owner)
                        ast_string_field_build(p->owner, call_forward, "SIP/%s", s);
@@ -11335,6 +11353,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
        }
 
        if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) { /* This call is already gone */
+               if (option_debug)
                ast_log(LOG_DEBUG, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
                return;
        }
@@ -11653,8 +11672,10 @@ static int handle_response_register(struct sip_pvt *p, int resp, char *rest, str
                r->regtime = time(NULL);                /* Reset time of last succesful registration */
                manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelDriver: SIP\r\nDomain: %s\r\nStatus: %s\r\n", r->hostname, regstate2str(r->regstate));
                r->regattempts = 0;
+               if (option_debug)
                ast_log(LOG_DEBUG, "Registration successful\n");
                if (r->timeout > -1) {
+                       if (option_debug)
                        ast_log(LOG_DEBUG, "Cancelling timeout %d\n", r->timeout);
                        ast_sched_del(sched, r->timeout);
                }
@@ -11934,6 +11955,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
                        if (sipmethod == SIP_INVITE)
                                handle_response_invite(p, resp, rest, req, seqno);
                        else {
+                               if (option_debug)
                                ast_log(LOG_DEBUG, "Got 491 on %s, unspported. Call ID %s\n", sip_methods[sipmethod].text, p->callid);
                                ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
                        }
@@ -11995,6 +12017,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
                                        is yet another failure of not having a layer 2 (again, YAY
                                         IETF for thinking ahead).  So we treat this as a call
                                         forward and hope we end up at the right place... */
+                                       if (option_debug)
                                        ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n");
                                        if (p->owner)
                                                ast_string_field_build(p->owner, call_forward,
@@ -12061,6 +12084,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
                        if (sipmethod == SIP_INVITE) {
                                handle_response_invite(p, resp, rest, req, seqno);
                        } else if (sipmethod == SIP_CANCEL) {
+                               if (option_debug)
                                ast_log(LOG_DEBUG, "Got 200 OK on CANCEL\n");
 
                                /* Wait for 487, then destroy */
@@ -12286,10 +12310,14 @@ static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct
        }
        ast_channel_unlock(transferer);
        if (!transferer || !transferee) {
-               if (!transferer)
+               if (!transferer) { 
+                       if (option_debug)
                        ast_log(LOG_DEBUG, "No transferer channel, giving up parking\n");
-               if (!transferee)
+               }
+               if (!transferee) {
+                       if (option_debug)
                        ast_log(LOG_DEBUG, "No transferee channel, giving up parking\n");
+               }
                return -1;
        }
        if ((d = ast_calloc(1, sizeof(*d)))) {
@@ -12925,6 +12953,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                                }
                        } else {
                                p->jointcapability = p->capability;
+                               if (option_debug)
                                ast_log(LOG_DEBUG, "Hm....  No sdp for the moment\n");
                        }
                        if (recordhistory) /* This is a response, note what it was for */
@@ -13326,6 +13355,7 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
                append_history(transferer, "Xfer", "Refer succeeded");
                transferer->refer->status = REFER_200OK;
                if (targetcall_pvt->owner) {
+                       if (option_debug)
                        ast_log(LOG_DEBUG, "SIP attended transfer: Unlocking channel %s\n", targetcall_pvt->owner->name);
                        ast_channel_unlock(targetcall_pvt->owner);
                }
@@ -13454,13 +13484,13 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
                case -2:        /* Syntax error */
                        transmit_response(p, "400 Bad Request (Refer-to missing)", req);
                        append_history(p, "Xfer", "Refer failed. Refer-to missing.");
-                       if (ast_test_flag(req, SIP_PKT_DEBUG))
+                       if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug)
                                ast_log(LOG_DEBUG, "SIP transfer to black hole can't be handled (no refer-to: )\n");
                        break;
                case -3:
                        transmit_response(p, "603 Declined (Non sip: uri)", req);
                        append_history(p, "Xfer", "Refer failed. Non SIP uri");
-                       if (ast_test_flag(req, SIP_PKT_DEBUG))
+                       if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug)
                                ast_log(LOG_DEBUG, "SIP transfer to non-SIP uri denied\n");
                        break;
                default:
@@ -13469,7 +13499,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
                        append_history(p, "Xfer", "Refer failed. Bad extension.");
                        transmit_notify_with_sipfrag(p, seqno, "404 Not found", TRUE);
                        ast_clear_flag(&p->flags[0], SIP_GOTREFER);     
-                       if (ast_test_flag(req, SIP_PKT_DEBUG))
+                       if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug)
                                ast_log(LOG_DEBUG, "SIP transfer to bad extension: %s\n", p->refer->refer_to);
                        break;
                } 
@@ -13827,15 +13857,18 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
                        /* For transfers, this could happen, but since we haven't seen it happening, let us just refuse this */
                        transmit_response(p, "403 Forbidden (within dialog)", req);
                        /* Do not destroy session, since we will break the call if we do */
+                       if (option_debug)
                        ast_log(LOG_DEBUG, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text);
                        return 0;
                } else if (ast_test_flag(req, SIP_PKT_DEBUG)) {
+                       if (option_debug) {
                        if (resubscribe)
                                ast_log(LOG_DEBUG, "Got a re-subscribe on existing subscription %s\n", p->callid);
                        else
                                ast_log(LOG_DEBUG, "Got a new subscription %s (possibly with auth)\n", p->callid);
                }
        }
+       }
 
        /* Check if we have a global disallow setting on subscriptions. 
                if so, we don't have to check peer/user settings after auth, which saves a lot of processing
@@ -15116,6 +15149,7 @@ static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char
        if (ast_strlen_zero(configuration))
                return authlist;
 
+       if (option_debug)
        ast_log(LOG_DEBUG, "Auth config ::  %s\n", configuration);
 
        ast_copy_string(authcopy, configuration, sizeof(authcopy));