Author: rizzo
Date: Fri Jul 20 12:08:17 2007
New Revision: 76061
URL: http://svn.digium.com/view/asterisk?view=rev&rev=76061
Log:
more ast_log -> ast_debug conversion
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=76061&r1=76060&r2=76061
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Fri Jul 20 12:08:17 2007
@@ -15220,14 +15220,14 @@
/* If we do not support SIP domains, all transfers are local */
if (allow_external_domains &&
check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
p->refer->localtransfer = 1;
- if (sipdebug && option_debug > 2)
- ast_log(LOG_DEBUG, "This SIP transfer is local : %s\n",
p->refer->refer_to_domain);
+ if (sipdebug)
+ ast_debug(3, "This SIP transfer is local : %s\n",
p->refer->refer_to_domain);
} else if (AST_LIST_EMPTY(&domain_list)) {
/* This PBX don't bother with SIP domains, so all transfers are
local */
p->refer->localtransfer = 1;
} else
- if (sipdebug && option_debug > 2)
- ast_log(LOG_DEBUG, "This SIP transfer is to a remote
SIP extension (remote domain %s)\n", p->refer->refer_to_domain);
+ if (sipdebug)
+ ast_debug(3, "This SIP transfer is to a remote SIP
extension (remote domain %s)\n", p->refer->refer_to_domain);
/* Is this a repeat of a current request? Ignore it */
/* Don't know what else to do right now. */
@@ -15265,15 +15265,15 @@
/* Find the other part of the bridge (2) - transferee */
current.chan2 = ast_bridged_channel(current.chan1);
- if (sipdebug && option_debug > 2)
- ast_log(LOG_DEBUG, "SIP %s transfer: Transferer channel %s,
transferee channel %s\n", p->refer->attendedtransfer ? "attended" : "blind",
current.chan1->name, current.chan2 ? current.chan2->name : "<none>");
+ if (sipdebug)
+ ast_debug(3, "SIP %s transfer: Transferer channel %s,
transferee channel %s\n", p->refer->attendedtransfer ? "attended" : "blind",
current.chan1->name, current.chan2 ? current.chan2->name : "<none>");
if (!current.chan2 && !p->refer->attendedtransfer) {
/* No bridged channel, propably IVR or echo or similar... */
/* Guess we should masquerade or something here */
/* Until we figure it out, refuse transfer of such calls */
- if (sipdebug && option_debug > 2)
- ast_log(LOG_DEBUG,"Refused SIP transfer on non-bridged
channel.\n");
+ if (sipdebug)
+ ast_debug(3,"Refused SIP transfer on non-bridged
channel.\n");
p->refer->status = REFER_FAILED;
append_history(p, "Xfer", "Refer failed. Non-bridged channel.");
transmit_response(p, "603 Declined", req);
@@ -15281,8 +15281,8 @@
}
if (current.chan2) {
- if (sipdebug && option_debug > 3)
- ast_log(LOG_DEBUG, "Got SIP transfer, applying to
bridged peer '%s'\n", current.chan2->name);
+ if (sipdebug)
+ ast_debug(4, "Got SIP transfer, applying to bridged
peer '%s'\n", current.chan2->name);
ast_queue_control(current.chan1, AST_CONTROL_UNHOLD);
}
@@ -15294,8 +15294,8 @@
if ((res = local_attended_transfer(p, ¤t, req, seqno)))
return res; /* We're done with the transfer */
/* Fall through for remote transfers that we did not find
locally */
- if (sipdebug && option_debug > 3)
- ast_log(LOG_DEBUG, "SIP attended transfer: Still not
our call - generating INVITE with replaces\n");
+ if (sipdebug)
+ ast_debug(4, "SIP attended transfer: Still not our call
- generating INVITE with replaces\n");
/* Fallthrough if we can't find the call leg internally */
}
@@ -15309,8 +15309,8 @@
ast_clear_flag(&p->flags[0], SIP_GOTREFER);
p->refer->status = REFER_200OK;
append_history(p, "Xfer", "REFER to call parking.");
- if (sipdebug && option_debug > 3)
- ast_log(LOG_DEBUG, "SIP transfer to parking: trying to
park %s. Parked by %s\n", current.chan2->name, current.chan1->name);
+ if (sipdebug)
+ ast_debug(4, "SIP transfer to parking: trying to park
%s. Parked by %s\n", current.chan2->name, current.chan1->name);
sip_park(current.chan2, current.chan1, req, seqno);
return res;
}
@@ -15319,8 +15319,7 @@
transmit_response(p, "202 Accepted", req);
if (current.chan1 && current.chan2) {
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "chan1->name: %s\n",
current.chan1->name);
+ ast_debug(3, "chan1->name: %s\n", current.chan1->name);
pbx_builtin_setvar_helper(current.chan1, "BLINDTRANSFER",
current.chan2->name);
}
if (current.chan2) {
@@ -15382,8 +15381,7 @@
if (!res) {
/* Success - we have a new channel */
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "%s transfer succeeded. Telling
transferer.\n", p->refer->attendedtransfer? "Attended" : "Blind");
+ ast_debug(3, "%s transfer succeeded. Telling transferer.\n",
p->refer->attendedtransfer? "Attended" : "Blind");
transmit_notify_with_sipfrag(p, seqno, "200 Ok", TRUE);
if (p->refer->localtransfer)
p->refer->status = REFER_200OK;
@@ -15396,8 +15394,7 @@
res = 0;
} else {
ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
/* Don't delay hangup */
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "%s transfer failed. Resuming
original call.\n", p->refer->attendedtransfer? "Attended" : "Blind");
+ ast_debug(3, "%s transfer failed. Resuming original call.\n",
p->refer->attendedtransfer? "Attended" : "Blind");
append_history(p, "Xfer", "Refer failed.");
/* Failure of some kind */
p->refer->status = REFER_FAILED;
@@ -15419,8 +15416,7 @@
if (p->owner && p->owner->_state == AST_STATE_UP) {
/* This call is up, cancel is ignored, we need a bye */
transmit_response(p, "200 OK", req);
- if (option_debug)
- ast_log(LOG_DEBUG, "Got CANCEL on an answered call.
Ignoring... \n");
+ ast_debug(1, "Got CANCEL on an answered call. Ignoring... \n");
return 0;
}
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as
applicable */
@@ -15515,8 +15511,8 @@
p->invitestate = INV_TERMINATED;
copy_request(&p->initreq, req);
- if (sipdebug && option_debug)
- ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid
%s\n", sip_methods[req->method].text, p->callid);
+ if (sipdebug)
+ ast_debug(1, "Initializing initreq for method %s - callid
%s\n", sip_methods[req->method].text, p->callid);
check_via(p, req);
sip_alreadygone(p);
@@ -15569,12 +15565,10 @@
}
} else if (p->owner) {
ast_queue_hangup(p->owner);
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Received bye, issuing owner
hangup\n");
+ ast_debug(3, "Received bye, issuing owner hangup\n");
} else {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Received bye, no owner,
selfdestruct soon.\n");
+ ast_debug(3, "Received bye, no owner, selfdestruct soon.\n");
}
transmit_response(p, "200 OK", req);
@@ -15612,16 +15606,13 @@
/* 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);
+ ast_debug(1, "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);
- }
+ if (resubscribe)
+ ast_debug(1, "Got a re-subscribe on existing
subscription %s\n", p->callid);
+ else
+ ast_debug(1, "Got a new subscription %s
(possibly with auth)\n", p->callid);
}
}
@@ -15640,16 +15631,15 @@
ast_verbose("Creating new subscription\n");
copy_request(&p->initreq, req);
- if (option_debug > 3 && sipdebug)
- ast_log(LOG_DEBUG, "Initializing initreq for method %s
- callid %s\n", sip_methods[req->method].text, p->callid);
+ if (sipdebug)
+ ast_debug(4, "Initializing initreq for method %s -
callid %s\n", sip_methods[req->method].text, p->callid);
check_via(p, req);
} else if (ast_test_flag(req, SIP_PKT_DEBUG) && req_ignore(req))
ast_verbose("Ignoring this SUBSCRIBE request\n");
/* Find parameters to Event: header value and remove them for now */
if (ast_strlen_zero(eventheader)) {
- if (option_debug > 1)
- ast_log(LOG_DEBUG, "Received SIP subscribe for unknown
event package: <none>\n");
+ ast_debug(2, "Received SIP subscribe for unknown event package:
<none>\n");
transmit_response(p, "489 Bad Event", req);
set_destroy(p);
return 0;
@@ -15759,8 +15749,7 @@
} else if (!strcmp(event, "message-summary")) {
if (!ast_strlen_zero(accept) && strcmp(accept,
"application/simple-message-summary")) {
/* Format requested that we do not support */
- if (option_debug > 1)
- ast_log(LOG_DEBUG, "Received SIP mailbox
subscription for unknown format: %s\n", accept);
+ ast_debug(2, "Received SIP mailbox subscription for
unknown format: %s\n", accept);
transmit_response(p, "406 Not Acceptable", req);
set_destroy(p);
unref_peer(authpeer);
@@ -15796,8 +15785,7 @@
p->relatedpeer = authpeer; /* Link from pvt to peer */
/* Do not release authpeer here */
} else { /* At this point, Asterisk does not understand the specified
event */
- if (option_debug > 1)
- ast_log(LOG_DEBUG, "Received SIP subscribe for unknown
event package: %s\n", event);
+ ast_debug(2, "Received SIP subscribe for unknown event package:
%s\n", event);
transmit_response(p, "489 Bad Event", req);
set_destroy(p);
unref_peer(authpeer);
@@ -15918,8 +15906,8 @@
/* Use this as the basis */
copy_request(&p->initreq, req);
- if (option_debug > 3 && sipdebug)
- ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid
%s\n", sip_methods[req->method].text, p->callid);
+ if (sipdebug)
+ ast_debug(4, "Initializing initreq for method %s - callid
%s\n", sip_methods[req->method].text, p->callid);
check_via(p, req);
if ((res = register_verify(p, sin, req, e)) < 0) {
const char *reason = "";
@@ -16004,8 +15992,7 @@
within an existing dialog */
/* Response to our request -- Do some sanity checks */
if (p->ocseq < seqno) {
- if (option_debug)
- ast_log(LOG_DEBUG, "Ignoring out of order
response %d (expecting %d)\n", seqno, p->ocseq);
+ ast_debug(1, "Ignoring out of order response %d
(expecting %d)\n", seqno, p->ocseq);
return -1;
} else if (p->ocseq != seqno) {
/* ignore means "don't do anything with it" but still
have to
@@ -16035,14 +16022,12 @@
*/
p->method = req->method; /* Find out which SIP method they are
using */
- if (option_debug > 3)
- ast_log(LOG_DEBUG, "**** Received %s (%d) - Command in SIP
%s\n", sip_methods[p->method].text, sip_methods[p->method].id, req->rlPart1);
+ ast_debug(4, "**** Received %s (%d) - Command in SIP %s\n",
sip_methods[p->method].text, sip_methods[p->method].id, req->rlPart1);
if (p->icseq == UNINITIALIZED_ICSEQ) { /* not initialized - anything
is good */
p->icseq = seqno;
} else if (seqno < p->icseq) { /* old packet */
- if (option_debug)
- ast_log(LOG_DEBUG, "Ignoring too old SIP packet packet
%d (expecting >= %d)\n", seqno, p->icseq);
+ ast_debug(1, "Ignoring too old SIP packet packet %d (expecting
>= %d)\n", seqno, p->icseq);
if (req->method != SIP_ACK)
transmit_response(p, "503 Server error", req); /* We
must respond according to RFC 3261 sec 12.2 */
return -1;
@@ -16054,8 +16039,7 @@
* side might have lost our message.
*/
ast_set_flag(req, SIP_PKT_IGNORE);
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Ignoring SIP message because of
retransmit (%s Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq,
seqno);
+ ast_debug(3, "Ignoring SIP message because of retransmit (%s
Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno);
} else {
/* Good sequence number - record it. It can be anything larger
than the
* previous sequence number, not necessarily incremented by 1.
@@ -16086,8 +16070,7 @@
transmit_response(p, "481 Call/Transaction Does
Not Exist", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else {
- if (option_debug)
- ast_log(LOG_DEBUG, "Got ACK for unknown
dialog... strange.\n");
+ ast_debug(1, "Got ACK for unknown dialog...
strange.\n");
}
return res;
}
@@ -16248,8 +16231,8 @@
}
}
- if (option_debug && res == buflen)
- ast_log(LOG_DEBUG, "Received packet exceeds buffer. Data is
possibly lost\n");
+ if (res == buflen)
+ ast_debug(1, "Received packet exceeds buffer. Data is possibly
lost\n");
req.len = res;
if(sip_debug_test_addr(&sin)) /* Set the debug flag early on packet
level */
ast_set_flag(&req, SIP_PKT_DEBUG);
@@ -16276,8 +16259,7 @@
p = find_call(&req, &sin, req.method); /* returns p locked */
ast_mark(prof_find, 0);
if (p == NULL) {
- if (option_debug)
- ast_log(LOG_DEBUG, "Invalid SIP message -
rejected , no callid, len %d\n", req.len);
+ ast_debug(1, "Invalid SIP message - rejected , no
callid, len %d\n", req.len);
return 1;
}
/* Go ahead and lock the owner if it has one -- we may need it
*/
@@ -16285,8 +16267,7 @@
if (!p->owner || !ast_channel_trylock(p->owner))
break; /* locking succeeded */
ast_verbose("loop %d p %p chan %p trylock failed\n", lockretry,
p, p->owner);
- if (option_debug)
- ast_log(LOG_DEBUG, "Failed to grab owner channel lock,
trying again. (SIP call %s)\n", p->callid);
+ ast_debug(1, "Failed to grab owner channel lock, trying again.
(SIP call %s)\n", p->callid);
sip_pvt_unlock(p);
p = pvt_unref(p); /* release the reference, no good
anymore */
/* Sleep for a very short amount of time */
@@ -16309,8 +16290,7 @@
nounlock = 0;
if (handle_incoming(p, &req, &sin, &recount, &nounlock)) {
/* Request failed */
- if (option_debug)
- ast_log(LOG_DEBUG, "SIP message could not be handled,
bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
+ ast_debug(1, "SIP message could not be handled, bad request:
%-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
}
if (p->owner && !nounlock)
@@ -16529,13 +16509,13 @@
if ((res < 0) || (res > 1000))
res = 1000;
res = ast_io_wait(io, res);
- if (option_debug && res > 20)
- ast_log(LOG_DEBUG, "chan_sip: ast_io_wait ran %d all at
once\n", res);
+ if (res > 20)
+ ast_debug(1, "chan_sip: ast_io_wait ran %d all at
once\n", res);
ast_mutex_lock(&monlock);
if (res >= 0) {
res = ast_sched_runq(sched);
- if (option_debug && res >= 20)
- ast_log(LOG_DEBUG, "chan_sip: ast_sched_runq
ran %d all at once\n", res);
+ if (res >= 20)
+ ast_debug(1, "chan_sip: ast_sched_runq ran %d
all at once\n", res);
}
ast_mutex_unlock(&monlock);
}
@@ -16700,8 +16680,7 @@
if ((tmp = strchr(host, '@')))
host = tmp + 1;
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Checking device state for peer %s\n", host);
+ ast_debug(3, "Checking device state for peer %s\n", host);
if ((p = find_peer(host, NULL, 1))) {
if (p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) {
@@ -16765,8 +16744,7 @@
*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find
codec to connect to host */
return NULL;
}
- if (option_debug)
- ast_log(LOG_DEBUG, "Asked to create a SIP channel with formats:
%s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat));
+ ast_debug(1, "Asked to create a SIP channel with formats: %s\n",
ast_getformatname_multiple(tmp, sizeof(tmp), oldformat));
if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out
of memory or socket error)\n", dest);
@@ -16798,8 +16776,7 @@
if (create_addr(p, host)) {
*cause = AST_CAUSE_UNREGISTERED;
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Cant create SIP call - target
device not registred\n");
+ ast_debug(3, "Cant create SIP call - target device not
registred\n");
sip_destroy(p);
return NULL;
}
@@ -17111,8 +17088,7 @@
if (ast_strlen_zero(configuration))
return authlist;
- if (option_debug)
- ast_log(LOG_DEBUG, "Auth config :: %s\n", configuration);
+ ast_debug(1, "Auth config :: %s\n", configuration);
ast_copy_string(authcopy, configuration, sizeof(authcopy));
stringp = authcopy;
@@ -17381,8 +17357,7 @@
if (realtime) {
rpeerobjs++;
- if (option_debug > 2)
- ast_log(LOG_DEBUG,"-REALTIME- peer built. Name:
%s. Peer objects: %d\n", name, rpeerobjs);
+ ast_debug(3,"-REALTIME- peer built. Name: %s. Peer
objects: %d\n", name, rpeerobjs);
} else
speerobjs++;
ASTOBJ_INIT(peer);
@@ -17540,8 +17515,7 @@
if ((nowtime - regseconds) > 0) {
destroy_association(peer);
memset(&peer->addr, 0, sizeof(peer->addr));
- if (option_debug)
- ast_log(LOG_DEBUG, "Bah, we're expired
(%d/%d/%d)!\n", (int)(nowtime - regseconds), (int)regseconds, (int)nowtime);
+ ast_debug(1, "Bah, we're expired (%d/%d/%d)!\n",
(int)(nowtime - regseconds), (int)regseconds, (int)nowtime);
}
}
ast_copy_flags(&peer->flags[0], &peerflags[0], mask[0].flags);
@@ -17829,8 +17803,8 @@
if (context)
*context++ = '\0';
- if (option_debug && ast_strlen_zero(context))
- ast_log(LOG_DEBUG, "No context specified at
line %d for domain '%s'\n", v->lineno, domain);
+ if (ast_strlen_zero(context))
+ ast_debug(1, "No context specified at line %d
for domain '%s'\n", v->lineno, domain);
if (ast_strlen_zero(domain))
ast_log(LOG_WARNING, "Empty domain specified at
line %d\n", v->lineno);
else
@@ -18111,14 +18085,10 @@
memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
if (!p->pendinginvite) {
- if (option_debug > 2) {
- ast_log(LOG_DEBUG, "Sending reinvite on SIP
'%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid,
ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip.sin_addr), udptl ?
ntohs(p->udptlredirip.sin_port) : 0);
- }
+ ast_debug(3, "Sending reinvite on SIP '%s' - It's UDPTL
soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ?
p->udptlredirip.sin_addr : p->ourip.sin_addr), udptl ?
ntohs(p->udptlredirip.sin_port) : 0);
transmit_reinvite_with_sdp(p, TRUE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- if (option_debug > 2) {
- ast_log(LOG_DEBUG, "Deferring reinvite on SIP
'%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid,
ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip.sin_addr), udptl ?
ntohs(p->udptlredirip.sin_port) : 0);
- }
+ ast_debug(3, "Deferring reinvite on SIP '%s' - It's
UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ?
p->udptlredirip.sin_addr : p->ourip.sin_addr), udptl ?
ntohs(p->udptlredirip.sin_port) : 0);
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
@@ -18165,20 +18135,16 @@
}
if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
if (!p->pendinginvite) {
- if (option_debug > 2) {
- if (flag)
- ast_log(LOG_DEBUG, "Sending
reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid,
ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
- else
- ast_log(LOG_DEBUG, "Sending
reinvite on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid,
ast_inet_ntoa(p->ourip.sin_addr));
- }
+ if (flag)
+ ast_debug(3, "Sending reinvite on SIP
'%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid,
ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
+ else
+ ast_debug(3, "Sending reinvite on SIP
'%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid,
ast_inet_ntoa(p->ourip.sin_addr));
transmit_reinvite_with_sdp(p, TRUE);
} else if (!ast_test_flag(&p->flags[0],
SIP_PENDINGBYE)) {
- if (option_debug > 2) {
- if (flag)
- ast_log(LOG_DEBUG, "Deferring
reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid,
ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
- else
- ast_log(LOG_DEBUG, "Deferring
reinvite on SIP '%s' - It's UDPTL will be redirected to us (IP %s)\n",
p->callid, ast_inet_ntoa(p->ourip.sin_addr));
- }
+ if (flag)
+ ast_debug(3, "Deferring reinvite on SIP
'%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid,
ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
+ else
+ ast_debug(3, "Deferring reinvite on SIP
'%s' - It's UDPTL will be redirected to us (IP %s)\n", p->callid,
ast_inet_ntoa(p->ourip.sin_addr));
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
@@ -18189,12 +18155,10 @@
} else {
memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
}
- if (option_debug > 2) {
- if (flag)
- ast_log(LOG_DEBUG, "Responding 200 OK on SIP
'%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid,
ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
- else
- ast_log(LOG_DEBUG, "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));
- }
+ if (flag)
+ ast_debug(3, "Responding 200 OK on SIP '%s' - It's
UDPTL soon redirected to IP %s:%d\n", p->callid,
ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
+ else
+ 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) {
@@ -18339,17 +18303,12 @@
if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
if (chan->_state != AST_STATE_UP) { /* We are in early
state */
append_history(p, "ExtInv", "Initial invite sent with
remote bridge proposal.");
- if (option_debug)
- ast_log(LOG_DEBUG, "Early remote bridge setting
SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ?
p->redirip.sin_addr : p->ourip.sin_addr));
+ ast_debug(1, "Early remote bridge setting SIP '%s' -
Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr :
p->ourip.sin_addr));
} else if (!p->pendinginvite) { /* We are up, and have
no outstanding invite */
- if (option_debug > 2) {
- ast_log(LOG_DEBUG, "Sending reinvite on SIP
'%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ?
p->redirip.sin_addr : p->ourip.sin_addr));
- }
+ ast_debug(3, "Sending reinvite on SIP '%s' - It's audio
soon redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr
: p->ourip.sin_addr));
transmit_reinvite_with_sdp(p, FALSE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- if (option_debug > 2) {
- ast_log(LOG_DEBUG, "Deferring reinvite on SIP
'%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp
? p->redirip.sin_addr : p->ourip.sin_addr));
- }
+ ast_debug(3, "Deferring reinvite on SIP '%s' - It's
audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ?
p->redirip.sin_addr : p->ourip.sin_addr));
/* We have a pending Invite. Send re-invite when we're
done with the invite */
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
}
@@ -18582,8 +18541,7 @@
/*! \brief Reload module */
static int sip_do_reload(enum channelreloadreason reason)
{
- if (option_debug > 3)
- ast_log(LOG_DEBUG, "--------------- SIP reload started\n");
+ ast_debug(4, "--------------- SIP reload started\n");
clear_realm_authentication(authl);
clear_sip_domains();
@@ -18594,8 +18552,7 @@
ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do {
ASTOBJ_RDLOCK(iterator);
if (iterator->register_pvt) {
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Destroying active SIP
dialog for registry [EMAIL PROTECTED]", iterator->username, iterator->hostname);
+ ast_debug(3, "Destroying active SIP dialog for registry
[EMAIL PROTECTED]", iterator->username, iterator->hostname);
/* This will also remove references to the registry */
iterator->register_pvt =
sip_destroy(iterator->register_pvt);
}
@@ -18604,16 +18561,14 @@
/* Then, actually destroy users and registry */
ASTOBJ_CONTAINER_DESTROYALL(&userl, sip_destroy_user);
- if (option_debug > 3)
- ast_log(LOG_DEBUG, "--------------- Done destroying user
list\n");
+ ast_debug(4, "--------------- Done destroying user list\n");
registry_destroy_all();
ASTOBJ_CONTAINER_MARKALL(&peerl);
reload_config(reason);
/* Prune peers who still are supposed to be deleted */
ASTOBJ_CONTAINER_PRUNE_MARKED(&peerl, sip_destroy_peer);
- if (option_debug > 3)
- ast_log(LOG_DEBUG, "--------------- Done destroying pruned
peers\n");
+ ast_debug(4, "--------------- Done destroying pruned peers\n");
sip_poke_all_peers(); /* Send qualify (OPTIONS) to all peers
*/
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