Author: rizzo
Date: Sun Jul 22 13:41:57 2007
New Revision: 76365
URL: http://svn.digium.com/view/asterisk?view=rev&rev=76365
Log:
Cleanup of flags used in struct sip_request, moving them to
individual variables. Apart from SIP_PKT_IGNORE which was used
a zillion times, the other two are used seldom.
On passing:
- move the arrays to the end of struct sip_request, so a (small)
buffer overflow is less likely to overwrite the other fields;
- note that the 'ignore' argument to handle_invite_replaces() is not
used and should be removed (will be done in a separate commit).
Nothing to backport in this change.
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL:
http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=76365&r1=76364&r2=76365
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sun Jul 22 13:41:57 2007
@@ -631,7 +631,14 @@
#define DEC_CALL_RINGING 2
#define INC_CALL_RINGING 3
-/*! \brief sip_request: The data grabbed from the UDP socket */
+/* SIP packet flags - note, they do NOT go in struct sip_pkt but
+ * in struct sip_request. */
+#define SIP_PKT_IGNORE (1 << 2) /*!< This is a
re-transmit, ignore it */
+
+/*! \brief sip_request: The data grabbed from the UDP socket
+ * data[] contains the packet itself, additional fields are set
+ * after parsing.
+ */
struct sip_request {
char *rlPart1; /*!< SIP Method Name or "SIP/2.0" protocol
version */
char *rlPart2; /*!< The Request URI or Response Status */
@@ -639,12 +646,14 @@
int headers; /*!< # of SIP Headers */
int method; /*!< Method of this request */
int lines; /*!< Body Content */
- unsigned int flags; /*!< SIP_PKT Flags for this packet */
+ unsigned int sdp_start; /*!< the line number where the SDP begins */
+ unsigned int sdp_end; /*!< the line number where the SDP ends */
+ char debug; /*!< print extra debugging if non zero */
+ char has_to_tag; /*!< non-zero if packet has To: tag */
+ char ignore; /*!< if non-zero This is a re-transmit, ignore
it */
char *header[SIP_MAX_HEADERS];
char *line[SIP_MAX_LINES];
char data[SIP_MAX_PACKET];
- unsigned int sdp_start; /*!< the line number where the SDP begins */
- unsigned int sdp_end; /*!< the line number where the SDP ends */
};
/*
@@ -830,10 +839,6 @@
SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE |
SIP_PAGE2_BUGGY_MWI | \
SIP_PAGE2_TEXTSUPPORT )
-/* SIP packet flags */
-#define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
-#define SIP_PKT_WITH_TOTAG (1 << 1) /*!< This packet has a to-tag */
-#define SIP_PKT_IGNORE (1 << 2) /*!< This is a
re-transmit, ignore it */
/* T.38 set of flags */
#define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0
(unset)*/
@@ -1831,7 +1836,7 @@
/* Use this as the basis */
copy_request(&p->initreq, req);
parse_request(&p->initreq);
- if (ast_test_flag(req, SIP_PKT_DEBUG))
+ if (req->debug)
ast_verbose("Initreq: %d headers, %d lines\n",
p->initreq.headers, p->initreq.lines);
}
@@ -4889,7 +4894,7 @@
in sip.conf
*/
if (gettag(req, "To", totag, sizeof(totag)))
- ast_set_flag(req, SIP_PKT_WITH_TOTAG); /* Used in
handle_request/response */
+ req->has_to_tag = 1; /* Used in
handle_request/response */
gettag(req, "From", fromtag, sizeof(fromtag));
tag = (req->method == SIP_RESPONSE) ? totag : fromtag;
@@ -9039,7 +9044,7 @@
} else {
ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT);
transmit_response(p, "100 Trying", req);
- if (!(res = check_auth(p, req, peer->name,
peer->secret, peer->md5secret, SIP_REGISTER, uri, XMIT_UNRELIABLE,
ast_test_flag(req, SIP_PKT_IGNORE)))) {
+ if (!(res = check_auth(p, req, peer->name,
peer->secret, peer->md5secret, SIP_REGISTER, uri, XMIT_UNRELIABLE,
req->ignore))) {
sip_cancel_destroy(p);
/* We have a successful registration attempt
with proper authentication,
@@ -9806,7 +9811,7 @@
replace_cid(p, rpid_num, calleridname);
do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT_ROUTE) );
- if (!(res = check_auth(p, req, user->name, user->secret,
user->md5secret, sipmethod, uri2, reliable, ast_test_flag(req,
SIP_PKT_IGNORE)))) {
+ if (!(res = check_auth(p, req, user->name, user->secret,
user->md5secret, sipmethod, uri2, reliable, req->ignore))) {
sip_cancel_destroy(p);
ast_copy_flags(&p->flags[0], &user->flags[0],
SIP_FLAGS_TO_COPY);
ast_copy_flags(&p->flags[1], &user->flags[1],
SIP_PAGE2_FLAGS_TO_COPY);
@@ -9932,7 +9937,7 @@
ast_string_field_free(p, peersecret);
ast_string_field_free(p, peermd5secret);
}
- if (!(res = check_auth(p, req, peer->name, p->peersecret,
p->peermd5secret, sipmethod, uri2, reliable, ast_test_flag(req,
SIP_PKT_IGNORE)))) {
+ if (!(res = check_auth(p, req, peer->name, p->peersecret,
p->peermd5secret, sipmethod, uri2, reliable, req->ignore))) {
ast_copy_flags(&p->flags[0], &peer->flags[0],
SIP_FLAGS_TO_COPY);
ast_copy_flags(&p->flags[1], &peer->flags[1],
SIP_PAGE2_FLAGS_TO_COPY);
/* If we have a call limit, set flag */
@@ -12687,15 +12692,15 @@
switch (resp) {
case 100: /* Trying */
case 101: /* Dialog establishment */
- if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ if (!req->ignore)
sip_cancel_destroy(p);
check_pendings(p);
break;
case 180: /* 180 Ringing */
- if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ if (!req->ignore)
sip_cancel_destroy(p);
- if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
+ if (!req->ignore && p->owner) {
ast_queue_control(p->owner, AST_CONTROL_RINGING);
if (p->owner->_state != AST_STATE_UP) {
ast_setstate(p->owner, AST_STATE_RINGING);
@@ -12704,7 +12709,7 @@
if (find_sdp(req)) {
p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
- if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
+ if (!req->ignore && p->owner) {
/* Queue a progress frame only if we have SDP
in 180 */
ast_queue_control(p->owner,
AST_CONTROL_PROGRESS);
}
@@ -12713,13 +12718,13 @@
break;
case 183: /* Session progress */
- if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ if (!req->ignore)
sip_cancel_destroy(p);
/* Ignore 183 Session progress without SDP */
if (find_sdp(req)) {
p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
- if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
+ if (!req->ignore && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner,
AST_CONTROL_PROGRESS);
}
@@ -12728,11 +12733,11 @@
break;
case 200: /* 200 OK on invite - someone's answering our call */
- if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ if (!req->ignore)
sip_cancel_destroy(p);
p->authtries = 0;
if (find_sdp(req)) {
- if ((res = process_sdp(p, req)) && !ast_test_flag(req,
SIP_PKT_IGNORE))
+ if ((res = process_sdp(p, req)) && !req->ignore)
if (!reinvite)
/* This 200 OK's SDP is not acceptable,
so we need to ack, then hangup */
/* For re-invites, we try to recover */
@@ -12753,7 +12758,7 @@
should we care about resolving the
contact
or should we just send it?
*/
- if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ if (!req->ignore)
ast_set_flag(&p->flags[0],
SIP_PENDINGBYE);
}
@@ -12804,7 +12809,7 @@
ast_debug(1, "T38 changed state to %d on channel %s\n",
p->t38.state, p->owner ? p->owner->name : "<none>");
}
- if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
+ if (!req->ignore && p->owner) {
if (!reinvite) {
ast_queue_control(p->owner, AST_CONTROL_ANSWER);
if (global_callevents)
@@ -12818,7 +12823,7 @@
/* It's possible we're getting an 200 OK after we've
tried to disconnect
by sending CANCEL */
/* First send ACK, then send bye */
- if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ if (!req->ignore)
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* If I understand this right, the branch is different for a
non-200 ACK only */
@@ -12836,7 +12841,7 @@
/* Then we AUTH */
ast_string_field_free(p, theirtag); /* forget their old
tag, so we don't match tags when getting response */
- if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
+ if (!req->ignore) {
if (p->authtries < MAX_AUTHTRIES)
p->invitestate = INV_CALLING;
if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p,
req, resp, SIP_INVITE, 1)) {
@@ -12853,7 +12858,7 @@
/* First we ACK */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE,
FALSE);
ast_log(LOG_WARNING, "Received response: \"Forbidden\" from
'%s'\n", get_header(&p->initreq, "From"));
- if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner)
+ if (!req->ignore && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
p->needdestroy = 1;
sip_alreadygone(p);
@@ -12861,7 +12866,7 @@
case 404: /* Not found */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE,
FALSE);
- if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
+ if (p->owner && !req->ignore)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
sip_alreadygone(p);
break;
@@ -12880,10 +12885,10 @@
This transaction is already scheduled to be killed by
sip_hangup().
*/
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE,
FALSE);
- if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE)) {
+ if (p->owner && !req->ignore) {
ast_queue_hangup(p->owner);
append_history(p, "Hangup", "Got 487 on CANCEL request
from us. Queued AST hangup request");
- } else if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
+ } else if (!req->ignore) {
update_call_counter(p, DEC_CALL_LIMIT);
append_history(p, "Hangup", "Got 487 on CANCEL request
from us on call without owner. Killing this dialog.");
p->needdestroy = 1;
@@ -12907,12 +12912,12 @@
sides here?
*/
/* While figuring that out, hangup the call */
- if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
+ if (p->owner && !req->ignore)
ast_queue_control(p->owner,
AST_CONTROL_CONGESTION);
p->needdestroy = 1;
} else {
/* We can't set up this call, so give up */
- if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
+ if (p->owner && !req->ignore)
ast_queue_control(p->owner,
AST_CONTROL_CONGESTION);
p->needdestroy = 1;
}
@@ -12922,7 +12927,7 @@
/* We should support the retry-after at some point */
/* At this point, we treat this as a congestion */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE,
FALSE);
- if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
+ if (p->owner && !req->ignore)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
p->needdestroy = 1;
break;
@@ -13459,7 +13464,7 @@
p->needdestroy = 1;
} else if ((resp >= 100) && (resp < 200)) {
if (sipmethod == SIP_INVITE) {
- if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ if (!req->ignore)
sip_cancel_destroy(p);
if (find_sdp(req))
process_sdp(p, req);
@@ -13474,7 +13479,7 @@
} else {
/* Responses to OUTGOING SIP requests on INCOMING calls
get handled here. As well as out-of-call message responses */
- if (ast_test_flag(req, SIP_PKT_DEBUG))
+ if (req->debug)
ast_verbose("SIP Response message for INCOMING dialog
%s arrived\n", msg);
if (sipmethod == SIP_INVITE && resp == 200) {
@@ -13558,7 +13563,7 @@
default: /* Errors without handlers */
if ((resp >= 100) && (resp < 200)) {
if (sipmethod == SIP_INVITE) { /* re-invite */
- if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ if (!req->ignore)
sip_cancel_destroy(p);
}
}
@@ -14006,7 +14011,10 @@
}
/*! \brief Handle the transfer part of INVITE with a replaces: header,
- meaning a target pickup or an attended transfer */
+ meaning a target pickup or an attended transfer.
+ Used only once.
+ XXX 'ignore' is unused.
+ */
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req,
int debug, int ignore, int seqno, struct sockaddr_in *sin)
{
struct ast_frame *f;
@@ -14038,7 +14046,7 @@
else
ast_debug(4, "SIP transfer: Invite Replace incoming channel
should replace and hang up channel %s (one call leg)\n", replacecall->name);
- if (ast_test_flag(req, SIP_PKT_IGNORE)) {
+ if (req->ignore) {
ast_log(LOG_NOTICE, "Ignoring this INVITE with replaces in a
stupid way.\n");
/* We should answer something here. If we are here, the
call we are replacing exists, so an accepted
@@ -14209,7 +14217,7 @@
return 0;
}
- if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->pendinginvite) {
+ if (!req->ignore && p->pendinginvite) {
/* We already have a pending invite. Sorry. You are on hold. */
transmit_response(p, "491 Request Pending", req);
ast_debug(1, "Got INVITE on call where we already have pending
INVITE, deferring that - %s\n", p->callid);
@@ -14326,7 +14334,7 @@
/* Check if this is an INVITE that sets up a new dialog or
a re-invite in an existing dialog */
- if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
+ if (!req->ignore) {
int newcall = (p->initreq.headers ? TRUE : FALSE);
sip_cancel_destroy(p);
@@ -14364,7 +14372,7 @@
ast_verbose("Ignoring this INVITE request\n");
- if (!p->lastinvite && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner)
{
+ if (!p->lastinvite && !req->ignore && !p->owner) {
/* This is a new invite */
/* Handle authentication if this is our first invite */
res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin);
@@ -14464,7 +14472,7 @@
}
} else {
if (sipdebug) {
- if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ if (!req->ignore)
ast_debug(2, "Got a SIP re-invite for call
%s\n", p->callid);
else
ast_debug(2, "Got a SIP re-transmit of INVITE
for call %s\n", p->callid);
@@ -14473,14 +14481,14 @@
c = p->owner;
}
- if (!ast_test_flag(req, SIP_PKT_IGNORE) && p)
+ if (!req->ignore && p)
p->lastinvite = seqno;
if (replace_id) { /* Attended transfer or call pickup - we're the
target */
/* Go and take over the target call */
if (sipdebug)
ast_debug(4, "Sending this call to the invite/replcaes
handler %s\n", p->callid);
- return handle_invite_replaces(p, req, debug, ast_test_flag(req,
SIP_PKT_IGNORE), seqno, sin);
+ return handle_invite_replaces(p, req, debug, req->ignore,
seqno, sin);
}
@@ -14500,7 +14508,7 @@
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start
PBX :(\n");
p->invitestate = INV_COMPLETED;
- if (ast_test_flag(req, SIP_PKT_IGNORE))
+ if (req->ignore)
transmit_response(p, "503
Unavailable", req);
else
transmit_response_reliable(p,
"503 Unavailable", req);
@@ -14508,7 +14516,7 @@
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start
PBX (call limit reached) \n");
p->invitestate = INV_COMPLETED;
- if (ast_test_flag(req, SIP_PKT_IGNORE))
+ if (req->ignore)
transmit_response(p, "480
Temporarily Unavailable", req);
else
transmit_response_reliable(p,
"480 Temporarily Unavailable", req);
@@ -14532,7 +14540,7 @@
*nounlock = 1;
if (ast_pickup_call(c)) {
ast_log(LOG_NOTICE, "Nothing to pick up
for %s\n", p->callid);
- if (ast_test_flag(req, SIP_PKT_IGNORE))
+ if (req->ignore)
transmit_response(p, "503
Unavailable", req); /* OEJ - Right answer? */
else
transmit_response_reliable(p,
"503 Unavailable", req);
@@ -14581,7 +14589,7 @@
bridgepvt->t38.state = T38_DISABLED;
sip_pvt_unlock(bridgepvt);
ast_debug(2,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state,
bridgepeer->name);
- if
(ast_test_flag(req, SIP_PKT_IGNORE))
+ if (req->ignore)
transmit_response(p, "488 Not acceptable here", req);
else
transmit_response_reliable(p, "488 Not acceptable here", req);
@@ -14595,7 +14603,7 @@
}
} else {
/* Other side is not a SIP
channel */
- if (ast_test_flag(req,
SIP_PKT_IGNORE))
+ if (req->ignore)
transmit_response(p,
"488 Not acceptable here", req);
else
transmit_response_reliable(p, "488 Not acceptable here", req);
@@ -14625,7 +14633,7 @@
if (bridgepvt->t38.state ==
T38_ENABLED) {
ast_log(LOG_WARNING,
"RTP re-invite after T38 session not handled yet !\n");
/* Insted of this we
should somehow re-invite the other side of the bridge to RTP */
- if (ast_test_flag(req,
SIP_PKT_IGNORE))
+ if (req->ignore)
transmit_response(p, "488 Not Acceptable Here (unsupported)", req);
else
transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
@@ -14637,7 +14645,7 @@
/* Respond to normal re-invite */
if (sendok)
/* If this is not a re-invite or
something to ignore - it's critical */
- transmit_response_with_sdp(p, "200 OK",
req, (reinvite || ast_test_flag(req, SIP_PKT_IGNORE)) ? XMIT_UNRELIABLE :
XMIT_CRITICAL);
+ transmit_response_with_sdp(p, "200 OK",
req, (reinvite || req->ignore) ? XMIT_UNRELIABLE : XMIT_CRITICAL);
}
p->invitestate = INV_TERMINATED;
break;
@@ -14656,7 +14664,7 @@
ast_log(LOG_NOTICE, "Unable to create/find SIP
channel for this INVITE\n");
msg = "503 Unavailable";
}
- if (ast_test_flag(req, SIP_PKT_IGNORE))
+ if (req->ignore)
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
@@ -14835,7 +14843,7 @@
int res = 0;
- if (ast_test_flag(req, SIP_PKT_DEBUG))
+ if (req->debug)
ast_verbose("Call %s got a SIP call transfer from %s:
(REFER)!\n", p->callid, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "callee" :
"caller");
if (!p->owner) {
@@ -14843,7 +14851,7 @@
/* We can't handle that, so decline it */
ast_debug(3, "Call %s: Declined REFER, outside of dialog...\n",
p->callid);
transmit_response(p, "603 Declined (No dialog)", req);
- if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
+ if (!req->ignore) {
append_history(p, "Xfer", "Refer failed. Outside of
dialog.");
sip_alreadygone(p);
p->needdestroy = 1;
@@ -14861,7 +14869,7 @@
return 0;
}
- if(!ast_test_flag(req, SIP_PKT_IGNORE) && ast_test_flag(&p->flags[0],
SIP_GOTREFER)) {
+ if (!req->ignore && ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
/* Already have a pending REFER */
transmit_response(p, "491 Request pending", req);
append_history(p, "Xfer", "Refer failed. Request pending.");
@@ -14884,13 +14892,13 @@
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 (req->debug)
ast_debug(1, "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 (req->debug)
ast_debug(1, "SIP transfer to non-SIP uri
denied\n");
break;
default:
@@ -14899,7 +14907,7 @@
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 (req->debug)
ast_debug(1, "SIP transfer to bad extension:
%s\n", p->refer->refer_to);
break;
}
@@ -14922,7 +14930,7 @@
/* Is this a repeat of a current request? Ignore it */
/* Don't know what else to do right now. */
- if (ast_test_flag(req, SIP_PKT_IGNORE))
+ if (req->ignore)
return res;
/* If this is a blind transfer, we have the following
@@ -15219,7 +15227,7 @@
struct ast_channel *bridged_to;
/* If we have an INCOMING invite that we haven't answered, terminate
that transaction */
- if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) &&
!ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner)
+ if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) &&
!req->ignore && !p->owner)
transmit_response_reliable(p, "487 Request Terminated",
&p->initreq);
p->invitestate = INV_TERMINATED;
@@ -15295,8 +15303,8 @@
/*! \brief Handle incoming MESSAGE request */
static int handle_request_message(struct sip_pvt *p, struct sip_request *req)
{
- if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
- if (ast_test_flag(req, SIP_PKT_DEBUG))
+ if (!req->ignore) {
+ if (req->debug)
ast_verbose("Receiving message!\n");
receive_message(p, req);
} else
@@ -15325,7 +15333,7 @@
/* Do not destroy session, since we will break the call
if we do */
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)) {
+ } else if (req->debug) {
if (resubscribe)
ast_debug(1, "Got a re-subscribe on existing
subscription %s\n", p->callid);
else
@@ -15342,16 +15350,16 @@
return 0;
}
- if (!ast_test_flag(req, SIP_PKT_IGNORE) && !resubscribe) { /* Set
up dialog, new subscription */
+ if (!req->ignore && !resubscribe) { /* Set up dialog, new
subscription */
/* Use this as the basis */
- if (ast_test_flag(req, SIP_PKT_DEBUG))
+ if (req->debug)
ast_verbose("Creating new subscription\n");
copy_request(&p->initreq, req);
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) && ast_test_flag(req,
SIP_PKT_IGNORE))
+ } else if (req->debug && req->ignore)
ast_verbose("Ignoring this SUBSCRIBE request\n");
/* Find parameters to Event: header value and remove them for now */
@@ -15518,7 +15526,7 @@
p->stateid = ast_extension_state_add(p->context, p->exten,
cb_extensionstate, p);
}
- if (!ast_test_flag(req, SIP_PKT_IGNORE) && p)
+ if (!req->ignore && p)
p->lastinvite = seqno;
if (p && !p->needdestroy) {
p->expiry = atoi(get_header(req, "Expires"));
@@ -15705,7 +15713,7 @@
} else if (p->ocseq != seqno) {
/* ignore means "don't do anything with it" but still
have to
respond appropriately */
- ast_set_flag(req, SIP_PKT_IGNORE);
+ req->ignore = 1;
append_history(p, "Ignore", "Ignoring this
retransmit\n");
} else if (e) {
e = ast_skip_blanks(e);
@@ -15744,7 +15752,7 @@
/* ignore means "don't do anything with it" but still have to
respond appropriately. We do this if we receive a repeat of
the last sequence number */
- ast_set_flag(req, SIP_PKT_IGNORE);
+ req->ignore = 1;
ast_debug(3, "Ignoring SIP message because of retransmit (%s
Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno);
}
@@ -15768,9 +15776,9 @@
correct according to RFC 3261 */
/* Check if this a new request in a new dialog with a totag
already attached to it,
RFC 3261 - section 12.2 - and we don't want to mess
with recovery */
- if (!p->initreq.headers && ast_test_flag(req,
SIP_PKT_WITH_TOTAG)) {
+ if (!p->initreq.headers && req->has_to_tag) {
/* If this is a first request and it got a to-tag, it
is not for us */
- if (!ast_test_flag(req, SIP_PKT_IGNORE) && req->method
== SIP_INVITE) {
+ if (!req->ignore && req->method == SIP_INVITE) {
transmit_response_reliable(p, "481
Call/Transaction Does Not Exist", req);
/* Will cease to exist after ACK */
} else if (req->method != SIP_ACK) {
@@ -15816,9 +15824,9 @@
res = handle_request_register(p, req, sin, e);
break;
case SIP_INFO:
- if (ast_test_flag(req, SIP_PKT_DEBUG))
+ if (req->debug)
ast_verbose("Receiving INFO!\n");
- if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ if (!req->ignore)
handle_request_info(p, req);
else /* if ignoring, transmit response */
transmit_response(p, "200 OK", req);
@@ -15889,16 +15897,16 @@
req.data[res] = '\0';
req.len = res;
if(sip_debug_test_addr(&sin)) /* Set the debug flag early on packet
level */
- ast_set_flag(&req, SIP_PKT_DEBUG);
+ req.debug = 1;
if (pedanticsipchecking)
req.len = lws2sws(req.data, req.len); /* Fix multiline
headers */
- if (ast_test_flag(&req, SIP_PKT_DEBUG))
+ if (req.debug)
ast_verbose("\n<--- SIP read from %s:%d
--->\n%s\n<------------->\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port),
req.data);
parse_request(&req);
req.method = find_sip_method(req.rlPart1);
- if (ast_test_flag(&req, SIP_PKT_DEBUG))
+ if (req.debug)
ast_verbose("--- (%d headers %d lines)%s ---\n", req.headers,
req.lines, (req.headers + req.lines == 0) ? " Nat keepalive" : "");
if (req.headers < 2) /* Must have at least two headers */
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