Author: rizzo
Date: Fri Jul 20 16:40:06 2007
New Revision: 76212
URL: http://svn.digium.com/view/asterisk?view=rev&rev=76212
Log:
more ast_log --> ast_debug replacements, to reduce diff with trunk.
These actually change the original behaviour (e.g.
"option_debug || foo" becomes "option_debug && foo")
but it does not make a lot of difference in practice.
Modified:
team/rizzo/astobj2/channels/chan_sip.c
Modified: team/rizzo/astobj2/channels/chan_sip.c
URL:
http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=76212&r1=76211&r2=76212
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Fri Jul 20 16:40:06 2007
@@ -2150,6 +2150,7 @@
{
if (p->outboundproxy)
return &p->outboundproxy->ip;
+
return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? &p->recv
: &p->sa;
}
@@ -3705,8 +3706,7 @@
/* Decrement onhold count if applicable */
if (ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD) &&
global_notifyhold)
sip_peer_hold(fup, FALSE);
- if (option_debug > 1 || sipdebug)
- ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call
limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
+ ast_debug(2, "Call %s %s '%s' removed from call limit %d\n",
outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
break;
case INC_CALL_RINGING:
@@ -3731,8 +3731,8 @@
/* Continue */
ast_atomic_fetchadd_int(inuse, +1);
ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
- if (option_debug > 1 || sipdebug) {
- ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n",
outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
+ if (sipdebug) {
+ ast_debug(2, "Call %s %s '%s' is %d out of %d\n",
outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
}
break;
@@ -5715,11 +5715,11 @@
sin.sin_port = htons(udptlportno);
ast_udptl_set_peer(p->udptl, &sin);
if (debug) /* XXX really ? */
- ast_log(LOG_DEBUG,"Peer T.38 UDPTL is at port
%s:%d\n",ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
+ ast_debug(1, "Peer T.38 UDPTL is at port
%s:%d\n",ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
} else {
ast_udptl_stop(p->udptl);
if (debug)
- ast_log(LOG_DEBUG, "Peer doesn't provide T.38
UDPTL\n");
+ ast_debug(1, "Peer doesn't provide T.38
UDPTL\n");
}
}
@@ -5946,7 +5946,7 @@
p->t38.jointcapability |= (peert38capability &
p->t38.capability); /* Put the lower of our's and peer's speed */
}
if (debug)
- ast_log(LOG_DEBUG, "Our T38 capability = (%d), peer T38
capability (%d), joint T38 capability (%d)\n",
+ ast_debug(1, "Our T38 capability = (%d), peer T38
capability (%d), joint T38 capability (%d)\n",
p->t38.capability,
p->t38.peercapability,
p->t38.jointcapability);
@@ -6031,6 +6031,7 @@
if (debug)
ast_verbose("Peer video RTP is at port %s:%d\n",
ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
}
+
/* Setup text port number */
if (p->trtp && tsin.sin_port) {
ast_rtp_set_peer(p->trtp, &tsin);
@@ -6470,7 +6471,7 @@
if (p->route && !ast_strlen_zero(p->route->hop) &&
strstr(p->route->hop,";lr") == NULL) {
is_strict = TRUE;
if (sipdebug)
- ast_log(LOG_DEBUG, "Strict routing enforced for session
%s\n", p->callid);
+ ast_debug(1, "Strict routing enforced for session
%s\n", p->callid);
}
if (sipmethod == SIP_CANCEL)
@@ -6529,9 +6530,10 @@
add_header(req, "From", ot);
add_header(req, "To", of);
}
- /* Do not add Contact for BYE and Cancel requests */
+ /* Do not add Contact for MESSAGE, BYE and Cancel requests */
if (sipmethod != SIP_BYE && sipmethod != SIP_CANCEL && sipmethod !=
SIP_MESSAGE)
add_header(req, "Contact", p->our_contact);
+
copy_header(req, orig, "Call-ID");
add_header(req, "CSeq", tmp);
@@ -6898,13 +6900,13 @@
}
if (debug) /* XXX is the message correct ? */
- ast_log(LOG_DEBUG, "T.38 UDPTL is at %s port %d\n",
ast_inet_ntoa(p->ourip.sin_addr), ntohs(udptlsin.sin_port));
+ ast_debug(1, "T.38 UDPTL is at %s port %d\n",
ast_inet_ntoa(p->ourip.sin_addr), ntohs(udptlsin.sin_port));
/* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */
if (debug) {
- ast_log(LOG_DEBUG, "Our T38 capability (%d), peer T38
capability (%d), joint capability (%d)\n",
+ ast_debug(1, "Our T38 capability (%d), peer T38 capability
(%d), joint capability (%d)\n",
p->t38.capability,
p->t38.peercapability,
p->t38.jointcapability);
@@ -7735,7 +7737,7 @@
add_header(&req,
headdup, content);
if (sipdebug)
-
ast_log(LOG_DEBUG, "Adding SIP Header \"%s\" with content :%s: \n", headdup,
content);
+ ast_debug(1,
"Adding SIP Header \"%s\" with content :%s: \n", headdup, content);
}
}
}
@@ -8260,7 +8262,7 @@
* do not like the old nonce (for good reasons) and challenge
us again.
*/
if (sipdebug)
- ast_log(LOG_DEBUG, " >>> Re-using Auth data for
[EMAIL PROTECTED]", r->username, r->hostname);
+ ast_debug(1, " >>> Re-using Auth data for [EMAIL
PROTECTED]", r->username, r->hostname);
ast_string_field_set(p, realm, r->realm);
ast_string_field_set(p, nonce, r->nonce);
ast_string_field_set(p, domain, r->domain);
@@ -8340,8 +8342,8 @@
char *ttag, *ftag;
char *theirtag = ast_strdupa(p->theirtag);
- if (option_debug || sipdebug)
- ast_log(LOG_DEBUG, "SIP transfer of %s to %s\n", p->callid,
dest);
+ if (sipdebug)
+ ast_debug(1, "SIP transfer of %s to %s\n", p->callid, dest);
/* Are we transfering an inbound or outbound call ? */
if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
@@ -11991,17 +11993,17 @@
return;
}
- ast_log(LOG_DEBUG, "\n---------- SIP HISTORY for '%s' \n",
dialog->callid);
+ ast_debug(1, "\n---------- SIP HISTORY for '%s' \n", dialog->callid);
if (dialog->subscribed)
- ast_log(LOG_DEBUG, " * Subscription\n");
+ ast_debug(1, " * Subscription\n");
else
- ast_log(LOG_DEBUG, " * SIP Call\n");
+ ast_debug(1, " * SIP Call\n");
if (dialog->history)
AST_LIST_TRAVERSE(dialog->history, hist, list)
- ast_log(LOG_DEBUG, " %-3.3d. %s\n", ++x, hist->event);
+ ast_debug(1, " %-3.3d. %s\n", ++x, hist->event);
if (!x)
- ast_log(LOG_DEBUG, "Call '%s' has no history\n",
dialog->callid);
- ast_log(LOG_DEBUG, "\n---------- END SIP HISTORY for '%s' \n",
dialog->callid);
+ ast_debug(1, "Call '%s' has no history\n", dialog->callid);
+ ast_debug(1, "\n---------- END SIP HISTORY for '%s' \n",
dialog->callid);
}
@@ -12453,7 +12455,7 @@
secret = auth->secret;
md5secret = auth->md5secret;
if (sipdebug)
- ast_log(LOG_DEBUG,"Using realm %s authentication for
call %s\n", p->realm, p->callid);
+ ast_debug(1, "Using realm %s authentication for call
%s\n", p->realm, p->callid);
} else {
/* No authentication, use credentials in the dialog
* (coming from user/peer/registration)
@@ -13835,7 +13837,7 @@
} else if (sipmethod == SIP_BYE) {
set_destroy(p);
} else if (sipdebug) {
- ast_log(LOG_DEBUG, "Remote host can't match
request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
+ ast_debug(1, "Remote host can't match request
%s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
}
break;
case 501: /* Not Implemented */
@@ -14118,7 +14120,7 @@
ast_log(LOG_WARNING, "Failed to masquerade %s into
%s\n", peerb->name, peerc->name);
res = -1;
} else
- ast_log(LOG_DEBUG, "SIP transfer: Succeeded to
masquerade channels.\n");
+ ast_debug(4, "SIP transfer: Succeeded to masquerade
channels.\n");
return res;
} else {
ast_log(LOG_NOTICE, "SIP Transfer attempted with no appropriate
bridged calls to transfer\n");
@@ -14126,7 +14128,7 @@
ast_softhangup_nolock(transferer->chan1,
AST_SOFTHANGUP_DEV);
if (target->chan1)
ast_softhangup_nolock(target->chan1,
AST_SOFTHANGUP_DEV);
- return -1;
+ return -1; /* XXX -2 in trunk ? */
}
return 0;
}
@@ -14378,6 +14380,7 @@
/* Answer the incoming call and set channel to UP state */
transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE);
+
ast_setstate(c, AST_STATE_UP);
/* Stop music on hold and other generators */
@@ -15025,22 +15028,20 @@
if (!target.chan2 || !(target.chan2->_state == AST_STATE_UP ||
target.chan2->_state == AST_STATE_RINGING) ) {
/* Wrong state of new channel */
- if (option_debug > 3) {
- if (target.chan2)
- ast_log(LOG_DEBUG, "SIP attended transfer:
Error: Wrong state of target call: %s\n", ast_state2str(target.chan2->_state));
- else if (target.chan1->_state != AST_STATE_RING)
- ast_log(LOG_DEBUG, "SIP attended transfer:
Error: No target channel\n");
- else
- ast_log(LOG_DEBUG, "SIP attended transfer:
Attempting transfer in ringing state\n");
- }
+ if (target.chan2)
+ ast_debug(4, "SIP attended transfer: Error: Wrong state
of target call: %s\n", ast_state2str(target.chan2->_state));
+ else if (target.chan1->_state != AST_STATE_RING)
+ ast_debug(4, "SIP attended transfer: Error: No target
channel\n");
+ else
+ ast_debug(4, "SIP attended transfer: Attempting
transfer in ringing state\n");
}
/* Transfer */
- if (option_debug > 3 && sipdebug) {
+ if (sipdebug) {
if (current->chan2) /* We have two bridges */
- ast_log(LOG_DEBUG, "SIP attended transfer: trying to
bridge %s and %s\n", target.chan1->name, current->chan2->name);
+ ast_debug(4, "SIP attended transfer: trying to bridge
%s and %s\n", target.chan1->name, current->chan2->name);
else /* One bridge, propably transfer of
IVR/voicemail etc */
- ast_log(LOG_DEBUG, "SIP attended transfer: trying to
make %s take over (masq) %s\n", target.chan1->name, current->chan1->name);
+ ast_debug(4, "SIP attended transfer: trying to make %s
take over (masq) %s\n", target.chan1->name, current->chan1->name);
}
ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /*
Delay hangup */
@@ -15807,11 +15808,11 @@
if (p->expiry < min_expiry && p->expiry > 0)
p->expiry = min_expiry;
- if (sipdebug || option_debug > 1) {
+ if (sipdebug) {
if (p->subscribed == MWI_NOTIFICATION && p->relatedpeer)
- ast_log(LOG_DEBUG, "Adding subscription for
mailbox notification - peer %s Mailbox %s\n", p->relatedpeer->name,
p->relatedpeer->mailbox);
+ ast_debug(2, "Adding subscription for mailbox
notification - peer %s Mailbox %s\n", p->relatedpeer->name,
p->relatedpeer->mailbox);
else
- ast_log(LOG_DEBUG, "Adding subscription for
extension %s context %s for peer %s\n", p->exten, p->context, p->username);
+ ast_debug(2, "Adding subscription for extension
%s context %s for peer %s\n", p->exten, p->context, p->username);
}
if (p->autokillid > -1)
sip_cancel_destroy(p); /* Remove subscription expiry
for renewals */
@@ -17038,7 +17039,7 @@
AST_LIST_UNLOCK(&domain_list);
if (sipdebug)
- ast_log(LOG_DEBUG, "Added local SIP domain '%s'\n", domain);
+ ast_debug(1, "Added local SIP domain '%s'\n", domain);
return 1;
}
@@ -18161,10 +18162,8 @@
ast_debug(3, "Responding 200 OK on SIP '%s' - It's
UDPTL soon redirected to us (IP %s)\n", p->callid,
ast_inet_ntoa(p->ourip.sin_addr));
pvt->t38.state = T38_ENABLED;
p->t38.state = T38_ENABLED;
- if (option_debug > 1) {
- ast_log(LOG_DEBUG, "T38 changed state to %d on channel
%s\n", pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>");
- ast_log(LOG_DEBUG, "T38 changed state to %d on channel
%s\n", p->t38.state, chan ? chan->name : "<none>");
- }
+ ast_debug(2, "T38 changed state to %d on channel %s\n",
pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>");
+ ast_debug(2, "T38 changed state to %d on channel %s\n",
p->t38.state, chan ? chan->name : "<none>");
transmit_response_with_t38_sdp(p, "200 OK", &p->initreq,
XMIT_CRITICAL);
}
p->lastrtprx = p->lastrtptx = time(NULL);
@@ -18414,7 +18413,7 @@
if (ok) {
pbx_builtin_setvar_helper (chan, varbuf, inbuf);
if (sipdebug)
- ast_log(LOG_DEBUG,"SIP Header added \"%s\" as %s\n",
inbuf, varbuf);
+ ast_debug(1, "SIP Header added \"%s\" as %s\n", inbuf,
varbuf);
} else {
ast_log(LOG_WARNING, "Too many SIP headers added, max 50\n");
}
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