[svn-commits] rizzo: branch 1.4 r44661 -
/branches/1.4/channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Oct 7 04:04:30 MST 2006
Author: rizzo
Date: Sat Oct 7 06:04:30 2006
New Revision: 44661
URL: http://svn.digium.com/view/asterisk?rev=44661&view=rev
Log:
merge rev 44253 - option_debug checking before printing to debug channel.
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?rev=44661&r1=44660&r2=44661&view=diff
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Sat Oct 7 06:04:30 2006
@@ -1574,7 +1574,7 @@
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 @@
((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 @@
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 @@
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;
}
@@ -3212,8 +3215,10 @@
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 @@
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 @@
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 @@
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 @@
/* 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 @@
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 @@
}
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 @@
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 @@
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 @@
}
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 @@
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 @@
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 @@
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 @@
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 @@
}
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 @@
}
} 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 @@
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 @@
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 @@
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;
}
@@ -13827,14 +13857,17 @@
/* 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.
@@ -15116,6 +15149,7 @@
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));
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