Author: rizzo
Date: Sun Jul 22 15:25:42 2007
New Revision: 76389
URL: http://svn.digium.com/view/asterisk?view=rev&rev=76389
Log:
merge from trunk
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=76389&r1=76388&r2=76389
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Sun Jul 22 15:25:42 2007
@@ -616,12 +616,6 @@
#define DEC_CALL_RINGING 2
#define INC_CALL_RINGING 3
-/* SIP packet flags, go into sip_request.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 */
-#define SIP_PKT_PARSED (1 << 3) /*!< lines are
NUL-separated */
-
/*!
* Incoming packet, or outgoing one (for the time being)
* For incoming packets, we first store the data from the socket in data[],
@@ -629,7 +623,7 @@
* Then call parse_request() and req.method = find_sip_method();
* to initialize the other fields. The \r\n at the end of line is
* replaced by \0, so that data[] is not a conforming one anymore.
- * flags has SIP_PKT_PARSED set to remember that we can run get_header()
+ * rlPart1 is set to remember that we can run get_header()
* on this kind of packet.
*
* For outgoing packets, we initialize the fields with init_req() or
init_resp()
@@ -644,10 +638,10 @@
* empty line, which we store as the last of the headers.
*
* parse_request() splits the first line as follows:
- * Requests have in the first line METHOD URI SIP/2.0
+ * Requests have in the first line method uri SIP/2.0
* rlPart1 = method; rlPart2 = uri;
- * Responses have in the first line SIP/2.0 code description
- * rlPart1 = SIP/2.0; rlPart2 = code + description;
+ * Responses have in the first line SIP/2.0 NNN description
+ * rlPart1 = SIP/2.0; rlPart2 = NNN + description;
*/
struct sip_request {
@@ -657,9 +651,11 @@
int headers; /*!< # of SIP Headers */
int method; /*!< Method of this request */
int lines; /*!< Body Content */
- unsigned int flags; /*!< SIP_PKT Flags for this packet */ /* XXX
should be ast_flags */
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];
@@ -679,12 +675,6 @@
char data[SIP_MAX_PACKET];
};
-/*! \brief test the IGNORE flags in a sip_request */
-static inline int req_ignore(struct sip_request *req)
-{
- return ast_test_flag(req, SIP_PKT_IGNORE);
-}
-
/*! \brief structure used in transfers */
struct sip_dual {
struct ast_channel *chan1; /*!< First channel involved */
@@ -751,11 +741,12 @@
};
/*--- Various flags for the flags field in the pvt structure
- Trying to sort these up:
- D: Dialog only
- DP: Dialog and peer/user
- P: Peer/user only, not dialog
- G: Global flag only
+ Trying to sort these up (one or more of the following):
+ D: Dialog
+ P: Peer/user
+ G: Global flag
+ When flags are used by multiple structures, it is important that
+ they have a common layout so it is easy to copy them.
*/
#define __SIP_ALREADYGONE (1 << 0) /*!< D: Whether or not
we've already been destroyed by our peer */
#define __SIP_NEEDDESTROY (1 << 1) /*!< D: if we need to
be destroyed by the monitor thread */
@@ -854,6 +845,7 @@
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP |
SIP_PAGE2_VIDEOSUPPORT | \
SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE |
SIP_PAGE2_BUGGY_MWI | \
SIP_PAGE2_TEXTSUPPORT )
+
/* T.38 set of flags */
#define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0
(unset)*/
@@ -1156,8 +1148,8 @@
int retrans; /*!< Retransmission number */
int method; /*!< SIP method for this packet
*/
int seqno; /*!< Sequence number */
- int is_resp; /*!< non-zero if this is a
response packet (e.g. 200 OK) */
- int is_fatal; /*!< non-zero if this is a
fatal error */
+ char is_resp; /*!< non-zero if this is a
response packet (e.g. 200 OK) */
+ char is_fatal; /*!< non-zero if this is a
fatal error */
struct sip_pvt *pvt; /*!< Owner AST call */
int retransid; /*!< Retransmission ID */
int timer_a; /*!< SIP timer A,
retransmission timer */
@@ -1782,10 +1774,10 @@
.answer = sip_answer, /* called with chan locked */
.read = sip_read, /* called with chan locked */
.write = sip_write, /* called with chan locked */
- .write_video = sip_write, /* called with chan locked */
- .write_text = sip_write, /* XXX */
+ .write_video = sip_write, /* called with chan locked */
+ .write_text = sip_write,
.indicate = sip_indicate, /* called with chan locked */
- .transfer = sip_transfer, /* called with chan locked */
+ .transfer = sip_transfer, /* called with chan locked */
.fixup = sip_fixup, /* called with chan locked */
.send_digit_begin = sip_senddigit_begin, /* called with chan
unlocked */
.send_digit_end = sip_senddigit_end,
@@ -4692,7 +4684,7 @@
{
int pass;
- if (!(req->flags & SIP_PKT_PARSED)) {
+ if (req->rlPart1 == NULL) {
ast_log(LOG_WARNING, "packet not parsed, cannot extract field
%s\n", name);
return "";
}
@@ -5154,7 +5146,7 @@
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));
arg.tag = (req->method == SIP_RESPONSE) ? totag : fromtag;
@@ -5378,7 +5370,7 @@
char *c = req->data, **dst = req->header;
int i = 0, lim = SIP_MAX_HEADERS - 1;
- if (req->flags & SIP_PKT_PARSED) {
+ if (req->rlPart1) {
ast_log(LOG_WARNING, "sorry, packet already parsed");
return;
}
@@ -5429,7 +5421,6 @@
ast_log(LOG_WARNING, "Too many lines, skipping <%s>\n", c);
/* Split up the first line parts */
determine_firstline_parts(req);
- req->flags |= SIP_PKT_PARSED;
}
/*!
@@ -9301,7 +9292,7 @@
} 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,
req_ignore(req)))) {
+ 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 succesful registration attempt
with proper authentication,
@@ -10091,7 +10082,7 @@
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, req_ignore(req)))) {
+ 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);
@@ -10222,7 +10213,7 @@
ast_string_field_free(p, secret);
ast_string_field_free(p, md5secret);
}
- if (!(res = check_auth(p, req, peer->name, p->secret, p->md5secret,
sipmethod, uri2, reliable, req_ignore(req)))) {
+ if (!(res = check_auth(p, req, peer->name, p->secret, p->md5secret,
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 */
@@ -11537,6 +11528,8 @@
msg = "Disabled, no localnet list";
else if (externip.sin_addr.s_addr == 0)
msg = "Disabled, externip is 0.0.0.0";
+ else if (stunaddr.sin_addr.s_addr != 0)
+ msg = "Enabled using STUN";
else if (!ast_strlen_zero(externhost))
msg = "Enabled using externhost";
else
@@ -12091,12 +12084,12 @@
} else {
ast_copy_string(buf, c, sizeof(buf));
}
+
if (!ast_strlen_zero((c = get_body(req, "Duration"))))
duration = atoi(c);
if (!duration)
duration = 100; /* 100 ms */
-
if (!p->owner) { /* not a PBX call */
transmit_response(p, "481 Call leg/transaction does not
exist", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -12180,29 +12173,17 @@
ast_unlock_call_features();
return;
}
- /* OEJ: Why is the DTMF code included in the record section? */
+ /* Send the feature code to the PBX as DTMF, just like the
handset had sent it */
f.len = 100;
for (j=0; j<strlen(feat->exten); j++) {
f.subclass = feat->exten[j];
ast_queue_frame(p->owner, &f);
if (sipdebug)
- ast_verbose("* DTMF-relay event received:
%c\n", f.subclass);
+ ast_verbose("* DTMF-relay event faked: %c\n",
f.subclass);
}
ast_unlock_call_features();
-#ifdef DISABLED_CODE
- /* And feat isn't used here - Is this code tested at all???
- We just send a reply ...
- */
- if (strcasecmp(c, "on")== 0) {
- ast_debug(1, "Got a Request to Record the channel!\n");
- transmit_response(p, "200 OK", req);
- return;
- } else if (strcasecmp(c, "off")== 0) {
- ast_debug(1, "Got a Request to Stop Recording the
channel\n");
- transmit_response(p, "200 OK", req);
- return;
- }
-#endif
+
+ ast_debug(1, "Got a Request to Record the channel, state %s\n",
c);
transmit_response(p, "200 OK", req);
return;
}
@@ -12997,15 +12978,15 @@
switch (resp) {
case 100: /* Trying */
case 101: /* Dialog establishment */
- if (!req_ignore(req))
+ if (!req->ignore)
sip_cancel_destroy(p);
check_pendings(p);
break;
case 180: /* 180 Ringing */
- if (!req_ignore(req))
+ if (!req->ignore)
sip_cancel_destroy(p);
- if (!req_ignore(req) && 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);
@@ -13014,7 +12995,7 @@
if (find_sdp(req)) {
p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
- if (!req_ignore(req) && 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);
}
@@ -13023,13 +13004,13 @@
break;
case 183: /* Session progress */
- if (!req_ignore(req))
+ 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 (!req_ignore(req) && p->owner) {
+ if (!req->ignore && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner,
AST_CONTROL_PROGRESS);
}
@@ -13038,11 +13019,11 @@
break;
case 200: /* 200 OK on invite - someone's answering our call */
- if (!req_ignore(req))
+ if (!req->ignore)
sip_cancel_destroy(p);
p->authtries = 0;
if (find_sdp(req)) {
- if ((res = process_sdp(p, req)) && !req_ignore(req))
+ 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 */
@@ -13063,7 +13044,7 @@
should we care about resolving the
contact
or should we just send it?
*/
- if (!req_ignore(req))
+ if (!req->ignore)
ast_set_flag(&p->flags[0],
SIP_PENDINGBYE);
}
@@ -13114,7 +13095,7 @@
ast_debug(1, "T38 changed state to %d on channel %s\n",
p->t38.state, p->owner ? p->owner->name : "<none>");
}
- if (!req_ignore(req) && p->owner) {
+ if (!req->ignore && p->owner) {
if (!reinvite) {
ast_queue_control(p->owner, AST_CONTROL_ANSWER);
if (global_callevents)
@@ -13128,7 +13109,7 @@
/* 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 (!req_ignore(req))
+ 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 */
@@ -13146,7 +13127,7 @@
/* Then we AUTH */
ast_string_field_free(p, theirtag); /* forget their old
tag, so we don't match tags when getting response */
- if (!req_ignore(req)) {
+ if (!req->ignore) {
/* XXX trunk has if (p->authtries < MAX_AUTHTRIES) ...
* but i am not sure if it is correct.
*/
@@ -13165,7 +13146,7 @@
/* First we ACK */
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 (!req_ignore(req) && p->owner)
+ if (!req->ignore && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
p->needdestroy = 1;
sip_alreadygone(p);
@@ -13173,7 +13154,7 @@
case 404: /* Not found */
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->owner && !req_ignore(req))
+ if (p->owner && !req->ignore)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
sip_alreadygone(p);
break;
@@ -13191,9 +13172,9 @@
This transaction is already scheduled to be killed by
sip_hangup().
*/
transmit_request(p, SIP_ACK, seqno, 0, 0);
- if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
+ if (p->owner && !req->ignore)
ast_queue_hangup(p->owner);
- else if (!ast_test_flag(req, SIP_PKT_IGNORE))
+ else if (!req->ignore)
update_call_counter(p, DEC_CALL_LIMIT);
break;
case 488: /* Not acceptable here */
@@ -13213,12 +13194,12 @@
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;
}
@@ -13229,7 +13210,7 @@
/* We should support the retry-after at some point */
/* At this point, we treat this as a congestion */
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->owner && !req_ignore(req))
+ if (p->owner && !req->ignore)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
p->needdestroy = 1;
break;
@@ -13819,7 +13800,7 @@
p->needdestroy = 1;
} else if ((resp >= 100) && (resp < 200)) {
if (sipmethod == SIP_INVITE) {
- if (!req_ignore(req))
+ if (!req->ignore)
sip_cancel_destroy(p);
if (find_sdp(req))
process_sdp(p, req);
@@ -13834,7 +13815,7 @@
} 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) {
@@ -13918,7 +13899,7 @@
default: /* Errors without handlers */
if ((resp >= 100) && (resp < 200)) {
if (sipmethod == SIP_INVITE) { /* re-invite */
- if (!req_ignore(req))
+ if (!req->ignore)
sip_cancel_destroy(p);
}
}
@@ -14402,7 +14383,7 @@
else
ast_debug(4, "SIP transfer: Invite Replace incoming channel
should replace and hang up channel %s (one call leg)\n", replacecall->name);
- if (req_ignore(req)) {
+ 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
@@ -14580,7 +14561,7 @@
return 0;
}
- if (!req_ignore(req) && 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);
@@ -14703,7 +14684,7 @@
/* Check if this is an INVITE that sets up a new dialog or
a re-invite in an existing dialog */
- if (req_ignore(req)) {
+ if (req->ignore) {
ast_verbose("Ignoring this INVITE request\n");
} else {
int newcall = (p->initreq.headers ? TRUE : FALSE);
@@ -14742,7 +14723,7 @@
}
}
- if (!p->lastinvite && !req_ignore(req) && !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);
@@ -14842,7 +14823,7 @@
}
} else {
if (sipdebug) {
- if (!req_ignore(req))
+ 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);
@@ -14850,14 +14831,14 @@
c = p->owner;
}
- if (!req_ignore(req) && 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, req_ignore(req),
seqno, sin);
+ return handle_invite_replaces(p, req, debug, req->ignore,
seqno, sin);
}
@@ -14877,7 +14858,7 @@
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start
PBX :(\n");
p->invitestate = INV_COMPLETED;
- if (req_ignore(req))
+ if (req->ignore)
transmit_response(p, "503
Unavailable", req);
else
transmit_response_reliable(p,
"503 Unavailable", req);
@@ -14885,7 +14866,7 @@
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start
PBX (call limit reached) \n");
p->invitestate = INV_COMPLETED;
- if (req_ignore(req))
+ if (req->ignore)
transmit_response(p, "480
Temporarily Unavailable", req);
else
transmit_response_reliable(p,
"480 Temporarily Unavailable", req);
@@ -14908,7 +14889,7 @@
ast_channel_unlock(c);
if (ast_pickup_call(c)) {
ast_log(LOG_NOTICE, "Nothing to pick up
for %s\n", p->callid);
- if (req_ignore(req))
+ if (req->ignore)
transmit_response(p, "503
Unavailable", req); /* OEJ - Right answer? */
else
transmit_response_reliable(p,
"503 Unavailable", req);
@@ -14957,7 +14938,7 @@
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
(req_ignore(req))
+ if (req->ignore)
transmit_response(p, "488 Not acceptable here", req);
else
transmit_response_reliable(p, "488 Not acceptable here", req);
@@ -14971,7 +14952,7 @@
}
} else {
/* Other side is not a SIP
channel */
- if (req_ignore(req))
+ if (req->ignore)
transmit_response(p,
"488 Not acceptable here", req);
else
transmit_response_reliable(p, "488 Not acceptable here", req);
@@ -15001,7 +14982,7 @@
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 (req_ignore(req))
+ if (req->ignore)
transmit_response(p, "488 Not Acceptable Here (unsupported)", req);
else
transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
@@ -15012,7 +14993,7 @@
}
/* Respond to normal re-invite */
if (sendok)
- transmit_response_with_sdp(p, "200 OK",
req, ast_test_flag(req, SIP_PKT_IGNORE) ? XMIT_UNRELIABLE : XMIT_CRITICAL);
+ transmit_response_with_sdp(p, "200 OK",
req, req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL);
}
p->invitestate = INV_TERMINATED;
break;
@@ -15031,7 +15012,7 @@
ast_log(LOG_NOTICE, "Unable to create/find SIP
channel for this INVITE\n");
msg = "503 Unavailable";
}
- if (req_ignore(req))
+ if (req->ignore)
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
@@ -15209,7 +15190,7 @@
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) {
@@ -15217,7 +15198,7 @@
/* 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 (!req_ignore(req)) {
+ if (!req->ignore) {
append_history(p, "Xfer", "Refer failed. Outside of
dialog.");
sip_alreadygone(p);
p->needdestroy = 1;
@@ -15235,7 +15216,7 @@
return 0;
}
- if(!req_ignore(req) && 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.");
@@ -15258,13 +15239,13 @@
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:
@@ -15273,7 +15254,7 @@
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;
}
@@ -15296,7 +15277,7 @@
/* Is this a repeat of a current request? Ignore it */
/* Don't know what else to do right now. */
- if (req_ignore(req))
+ if (req->ignore)
return res;
/* If this is a blind transfer, we have the following
@@ -15570,7 +15551,7 @@
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) &&
!req_ignore(req) && !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;
@@ -15646,8 +15627,8 @@
/*! \brief Handle incoming MESSAGE request */
static int handle_request_message(struct sip_pvt *p, struct sip_request *req)
{
- if (!req_ignore(req)) {
- if (ast_test_flag(req, SIP_PKT_DEBUG))
+ if (!req->ignore) {
+ if (req->debug)
ast_verbose("Receiving message!\n");
receive_message(p, req);
} else
@@ -15676,7 +15657,7 @@
/* 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
@@ -15693,22 +15674,22 @@
return 0;
}
- if (!req_ignore(req) && !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) && req_ignore(req))
+ } else if (req->debug && req->ignore)
ast_verbose("Ignoring this SUBSCRIBE request\n");
/* Find parameters to Event: header value and remove them for now */
if (ast_strlen_zero(eventheader)) {
+ transmit_response(p, "489 Bad Event", req);
ast_debug(2, "Received SIP subscribe for unknown event package:
<none>\n");
- transmit_response(p, "489 Bad Event", req);
p->needdestroy = 1;
return 0;
}
@@ -15809,6 +15790,7 @@
char mybuf[200];
snprintf(mybuf,sizeof(mybuf),"489 Bad Event (format
%s)", accept);
transmit_response(p, mybuf, req);
+
ast_log(LOG_WARNING,"SUBSCRIBE failure: unrecognized
format: '%s' pvt: subscribed: %d, stateid: %d, laststate: %d, dialogver: %d,
subscribecont: '%s', subscribeuri: '%s'\n",
accept, (int)p->subscribed, p->stateid,
p->laststate, p->dialogver, p->subscribecontext, p->subscribeuri);
p->needdestroy = 1;
@@ -15817,8 +15799,8 @@
} else if (!strcmp(event, "message-summary")) {
if (!ast_strlen_zero(accept) && strcmp(accept,
"application/simple-message-summary")) {
/* Format requested that we do not support */
+ transmit_response(p, "406 Not Acceptable", req);
ast_debug(2, "Received SIP mailbox subscription for
unknown format: %s\n", accept);
- transmit_response(p, "406 Not Acceptable", req);
p->needdestroy = 1;
unref_peer(authpeer);
return 0;
@@ -15864,7 +15846,7 @@
if (p->subscribed != MWI_NOTIFICATION && !resubscribe)
p->stateid = ast_extension_state_add(p->context, p->exten,
cb_extensionstate, p);
- if (!req_ignore(req) && p)
+ if (!req->ignore && p)
p->lastinvite = seqno;
if (p && !p->needdestroy) {
p->expiry = atoi(get_header(req, "Expires"));
@@ -16065,7 +16047,7 @@
} 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");
}
@@ -16106,7 +16088,7 @@
* required action), but still have to respond appropriately,
as the other
* side might have lost our message.
*/
- 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);
} else {
/* Good sequence number - record it. It can be anything larger
than the
@@ -16129,9 +16111,9 @@
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 (!req_ignore(req) && 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) {
@@ -16177,9 +16159,9 @@
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 (!req_ignore(req))
+ if (!req->ignore)
handle_request_info(p, req);
else /* if ignoring, transmit response */
transmit_response(p, "200 OK", req);
@@ -16302,11 +16284,11 @@
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);
+ if (sip_debug_test_addr(&sin)) /* Set the debug flag early on packet
level */
+ 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);
ast_mark(prof_parse, 1);
@@ -16314,7 +16296,7 @@
req.method = find_sip_method(req.rlPart1);
ast_mark(prof_parse, 0);
- 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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