[asterisk-commits] russell: trunk r99085 - in /trunk: ./ channels/ configs/ doc/ include/asteris...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jan 18 16:04:34 CST 2008
Author: russell
Date: Fri Jan 18 16:04:33 2008
New Revision: 99085
URL: http://svn.digium.com/view/asterisk?view=rev&rev=99085
Log:
Merge changes from team/group/sip-tcptls
This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
Added:
trunk/doc/siptls.txt
- copied unchanged from r99084, team/group/sip-tcptls/doc/siptls.txt
trunk/include/asterisk/tcptls.h
- copied unchanged from r99084, team/group/sip-tcptls/include/asterisk/tcptls.h
trunk/main/tcptls.c
- copied unchanged from r99084, team/group/sip-tcptls/main/tcptls.c
Modified:
trunk/CHANGES
trunk/CREDITS
trunk/channels/chan_sip.c
trunk/configs/sip.conf.sample
trunk/include/asterisk/http.h
trunk/main/Makefile
trunk/main/http.c
trunk/main/manager.c
Change Statistics:
trunk/CHANGES | 5
trunk/CREDITS | 7
trunk/channels/chan_sip.c | 791 +++++++++++++++++++++++++-----
trunk/configs/sip.conf.sample | 16
trunk/include/asterisk/http.h | 86 ---
trunk/main/Makefile | 2
trunk/main/http.c | 200 -------
trunk/main/manager.c | 3
8 files changed, 716 insertions(+), 394 deletions(-)
Modified: trunk/CHANGES
URL: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=diff&rev=99085&r1=99084&r2=99085
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Fri Jan 18 16:04:33 2008
@@ -128,6 +128,11 @@
SIP uri.
* Added a new global and per-peer option, qualifyfreq, which allows you to configure
the qualify frequency.
+ * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
+ were not properly torn down due to network or endpoint failures during an established
+ SIP session.
+ * Added TCP and TLS support for SIP. See doc/siptls.txt and configs/sip.conf.sample for
+ more information on how it is used.
IAX2 changes
------------
Modified: trunk/CREDITS
URL: http://svn.digium.com/view/asterisk/trunk/CREDITS?view=diff&rev=99085&r1=99084&r2=99085
==============================================================================
--- trunk/CREDITS (original)
+++ trunk/CREDITS Fri Jan 18 16:04:33 2008
@@ -56,7 +56,7 @@
and sip configs.
anthmct(AT)yahoo.com http://www.asterlink.com
-James Golovich - Innumerable contributions
+James Golovich - Innumerable contributions, including SIP TCP and TLS support.
You can find him and asterisk-perl at http://asterisk.gnuinter.net
Andre Bierwirth - Extension hints and status
@@ -175,6 +175,11 @@
Luigi Rizzo - astobj2, console_video, windows build, chan_oss cleanup,
and a bunch of infrastructure work (loader, new_cli, ...)
+
+Brett Bryant - digit option for musiconhold selection, ENUMQUERY and ENUMRESULT functions,
+ feature group configuration for features.conf, per-file CLI debug and verbose settings,
+ TCP and TLS support for SIP, and various bug fixes.
+ brettbryant(AT)gmail.com
=== OTHER CONTRIBUTIONS ===
John Todd - Monkey sounds and associated teletorture prompt
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=99085&r1=99084&r2=99085
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Jan 18 16:04:33 2008
@@ -175,6 +175,7 @@
#include "asterisk/translate.h"
#include "asterisk/version.h"
#include "asterisk/event.h"
+#include "asterisk/tcptls.h"
#ifndef FALSE
#define FALSE 0
@@ -574,8 +575,9 @@
/*! \brief SIP Extensions we support */
#define SUPPORTED_EXTENSIONS "replaces, timer"
-/*! \brief Standard SIP port from RFC 3261. DO NOT CHANGE THIS */
+/*! \brief Standard SIP and TLS port from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT 5060
+#define STANDARD_TLS_PORT 5061
/* Note: in many SIP headers, absence of a port number implies port 5060,
* and this is why we cannot change the above constant.
* There is a limited number of places in asterisk where we could,
@@ -757,7 +759,23 @@
#define DEC_CALL_RINGING 2
#define INC_CALL_RINGING 3
-/*! \brief The data grabbed from the UDP socket
+/*!< Define some SIP transports */
+enum sip_transport {
+ SIP_TRANSPORT_UDP = 1,
+ SIP_TRANSPORT_TCP = 1 << 1,
+ SIP_TRANSPORT_TLS = 1 << 2,
+};
+
+struct sip_socket {
+ ast_mutex_t *lock;
+ enum sip_transport type;
+ int fd;
+ uint16_t port;
+ struct server_instance *ser;
+};
+
+/*! \brief sip_request: The data grabbed from the UDP socket
+ *
* \verbatim
* Incoming messages: we first store the data from the socket in data[],
* adding a trailing \0 to make string parsing routines happy.
@@ -795,6 +813,7 @@
char *header[SIP_MAX_HEADERS];
char *line[SIP_MAX_LINES];
char data[SIP_MAX_PACKET];
+ struct sip_socket socket;
};
/*! \brief structure used in transfers */
@@ -1175,6 +1194,7 @@
AST_STRING_FIELD(rpid_from); /*!< Our RPID From header */
AST_STRING_FIELD(url); /*!< URL to be sent with next message to peer */
);
+ struct sip_socket socket;
unsigned int ocseq; /*!< Current outgoing seqno */
unsigned int icseq; /*!< Current incoming seqno */
ast_group_t callgroup; /*!< Call group */
@@ -1397,6 +1417,7 @@
struct sip_peer {
ASTOBJ_COMPONENTS(struct sip_peer); /*!< name, refcount, objflags, object pointers */
/*!< peer->name is the unique name of this object */
+ struct sip_socket socket;
char secret[80]; /*!< Password */
char md5secret[80]; /*!< Password in MD5 */
struct sip_auth *auth; /*!< Realm authentication list */
@@ -1500,6 +1521,7 @@
AST_STRING_FIELD(callback); /*!< Contact extension */
AST_STRING_FIELD(random);
);
+ enum sip_transport transport;
int portno; /*!< Optional port override */
int expire; /*!< Sched ID of expiration */
int expiry; /*!< Value to use for the Expires header */
@@ -1516,7 +1538,18 @@
char lastmsg[256]; /*!< Last Message sent/received */
};
+struct sip_threadinfo {
+ int stop;
+ pthread_t threadid;
+ struct server_instance *ser;
+ enum sip_transport type; /* We keep a copy of the type here so we can display it in the connection list */
+ AST_LIST_ENTRY(sip_threadinfo) list;
+};
+
/* --- Linked lists of various objects --------*/
+
+/*! \brief The thread list of TCP threads */
+static AST_LIST_HEAD_STATIC(threadl, sip_threadinfo);
/*! \brief The user list: Users and friends */
static struct ast_user_list {
@@ -1605,6 +1638,8 @@
*/
static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
+static int ourport_tcp;
+static int ourport_tls;
static struct sockaddr_in debugaddr;
static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
@@ -1637,6 +1672,10 @@
static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static int sip_senddigit_begin(struct ast_channel *ast, char digit);
static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration);
+
+static int handle_request_do(struct sip_request *req, struct sockaddr_in *sin);
+static int sip_standard_port(struct sip_socket s);
+static int sip_prepare_socket(struct sip_pvt *p);
/*--- Transmitting responses and requests */
static int sipsock_read(int *id, int fd, short events, void *ignore);
@@ -1811,6 +1850,11 @@
static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to debuglog at end of dialog, before destroying data */
static inline int sip_debug_test_addr(const struct sockaddr_in *addr);
static inline int sip_debug_test_pvt(struct sip_pvt *p);
+
+
+/*! \brief Append to SIP dialog history
+ \return Always returns 0 */
+#define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
static void sip_dump_history(struct sip_pvt *dialog);
@@ -1998,6 +2042,31 @@
*/
static struct ast_channel_tech sip_tech_info;
+static void *sip_tcp_worker_fn(void *);
+
+static struct ast_tls_config sip_tls_cfg;
+static struct ast_tls_config default_tls_cfg;
+
+static struct server_args sip_tcp_desc = {
+ .accept_fd = -1,
+ .master = AST_PTHREADT_NULL,
+ .tls_cfg = NULL,
+ .poll_timeout = -1,
+ .name = "sip tcp server",
+ .accept_fn = server_root,
+ .worker_fn = sip_tcp_worker_fn,
+};
+
+static struct server_args sip_tls_desc = {
+ .accept_fd = -1,
+ .master = AST_PTHREADT_NULL,
+ .tls_cfg = &sip_tls_cfg,
+ .poll_timeout = -1,
+ .name = "sip tls server",
+ .accept_fn = server_root,
+ .worker_fn = sip_tcp_worker_fn,
+};
+
/* wrapper macro to tell whether t points to one of the sip_tech descriptors */
#define IS_SIP_TECH(t) ((t) == &sip_tech || (t) == &sip_tech_info)
@@ -2038,6 +2107,119 @@
.get_codec = sip_get_codec,
};
+static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct server_instance *ser);
+
+static void *sip_tcp_helper_thread(void *data)
+{
+ struct sip_pvt *pvt = data;
+ struct server_instance *ser = pvt->socket.ser;
+
+ return _sip_tcp_helper_thread(pvt, ser);
+}
+
+static void *sip_tcp_worker_fn(void *data)
+{
+ struct server_instance *ser = data;
+
+ return _sip_tcp_helper_thread(NULL, ser);
+}
+
+/*! \brief SIP TCP helper function */
+static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct server_instance *ser)
+{
+ int res, cl;
+ struct sip_request req = { 0, } , reqcpy = { 0, };
+ struct sip_threadinfo *me;
+ char buf[1024];
+
+ me = ast_calloc(1, sizeof(*me));
+
+ if (!me)
+ goto cleanup2;
+
+ me->threadid = pthread_self();
+ me->ser = ser;
+ if (ser->ssl)
+ me->type = SIP_TRANSPORT_TLS;
+ else
+ me->type = SIP_TRANSPORT_TCP;
+
+ AST_LIST_LOCK(&threadl);
+ AST_LIST_INSERT_TAIL(&threadl, me, list);
+ AST_LIST_UNLOCK(&threadl);
+
+ req.socket.lock = ast_calloc(1, sizeof(*req.socket.lock));
+
+ if (!req.socket.lock)
+ goto cleanup;
+
+ ast_mutex_init(req.socket.lock);
+
+ for (;;) {
+ memset(req.data, 0, sizeof(req.data));
+ req.len = 0;
+ req.ignore = 0;
+
+ req.socket.fd = ser->fd;
+ if (ser->ssl) {
+ req.socket.type = SIP_TRANSPORT_TLS;
+ req.socket.port = htons(ourport_tls);
+ } else {
+ req.socket.type = SIP_TRANSPORT_TCP;
+ req.socket.port = htons(ourport_tcp);
+ }
+ res = ast_wait_for_input(ser->fd, -1);
+ if (res < 0) {
+ ast_log(LOG_DEBUG, "ast_wait_for_input returned %d\n", res);
+ goto cleanup;
+ }
+
+ /* Read in headers one line at a time */
+ while (req.len < 4 || strncmp((char *)&req.data + req.len - 4, "\r\n\r\n", 4)) {
+ if (req.socket.lock)
+ ast_mutex_lock(req.socket.lock);
+ if (!fgets(buf, sizeof(buf), ser->f))
+ goto cleanup;
+ if (req.socket.lock)
+ ast_mutex_unlock(req.socket.lock);
+ if (me->stop)
+ goto cleanup;
+ strncat(req.data, buf, sizeof(req.data) - req.len);
+ req.len = strlen(req.data);
+ }
+ parse_copy(&reqcpy, &req);
+ if (sscanf(get_header(&reqcpy, "Content-Length"), "%d", &cl)) {
+ while (cl > 0) {
+ if (req.socket.lock)
+ ast_mutex_lock(req.socket.lock);
+ if (!fread(buf, (cl < sizeof(buf)) ? cl : sizeof(buf), 1, ser->f))
+ goto cleanup;
+ if (req.socket.lock)
+ ast_mutex_unlock(req.socket.lock);
+ if (me->stop)
+ goto cleanup;
+ cl -= strlen(buf);
+ strncat(req.data, buf, sizeof(req.data) - req.len);
+ req.len = strlen(req.data);
+ }
+ }
+ req.socket.ser = ser;
+ handle_request_do(&req, &ser->requestor);
+ }
+
+cleanup:
+ AST_LIST_LOCK(&threadl);
+ AST_LIST_REMOVE(&threadl, me, list);
+ AST_LIST_UNLOCK(&threadl);
+ ast_free(me);
+cleanup2:
+ fclose(ser->f);
+ ast_free(ser);
+ ast_free(req.socket.lock);
+
+ return NULL;
+}
+
#define sip_pvt_lock(x) ast_mutex_lock(&x->pvt_lock)
#define sip_pvt_unlock(x) ast_mutex_unlock(&x->pvt_lock)
@@ -2076,10 +2258,6 @@
get_udptl_info: sip_get_udptl_peer,
set_udptl_peer: sip_set_udptl_peer,
};
-
-/*! \brief Append to SIP dialog history
- \return Always returns 0 */
-#define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
__attribute__ ((format (printf, 2, 3)));
@@ -2283,12 +2461,45 @@
return sip_debug_test_addr(sip_real_dst(p));
}
+static inline const char *get_transport(enum sip_transport t)
+{
+ switch (t) {
+ case SIP_TRANSPORT_UDP:
+ return "UDP";
+ case SIP_TRANSPORT_TCP:
+ return "TCP";
+ case SIP_TRANSPORT_TLS:
+ return "TLS";
+ }
+
+ return "UNKNOWN";
+}
+
/*! \brief Transmit SIP message */
static int __sip_xmit(struct sip_pvt *p, char *data, int len)
{
- int res;
+ int res = 0;
const struct sockaddr_in *dst = sip_real_dst(p);
- res = sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
+
+ ast_log(LOG_DEBUG, "Trying to put '%.10s' onto %s socket...\n", data, get_transport(p->socket.type));
+
+ if (sip_prepare_socket(p) < 0)
+ return XMIT_ERROR;
+
+ if (p->socket.lock)
+ ast_mutex_lock(p->socket.lock);
+
+ if (p->socket.type & SIP_TRANSPORT_UDP)
+ res = sendto(p->socket.fd, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
+ else {
+ if (p->socket.ser->f)
+ res = server_write(p->socket.ser, data, len);
+ else
+ ast_log(LOG_DEBUG, "No p->socket.ser->f len=%d\n", len);
+ }
+
+ if (p->socket.lock)
+ ast_mutex_unlock(p->socket.lock);
if (res == -1) {
switch (errno) {
@@ -2301,9 +2512,9 @@
}
if (res != len)
ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));
+
return res;
}
-
/*! \brief Build a Via header for a request */
static void build_via(struct sip_pvt *p)
@@ -2312,7 +2523,8 @@
const char *rport = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : "";
/* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */
- ast_string_field_build(p, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s",
+ ast_string_field_build(p, via, "SIP/2.0/%s %s:%d;branch=z9hG4bK%08x%s",
+ get_transport(p->socket.type),
ast_inet_ntoa(p->ourip.sin_addr),
ntohs(p->ourip.sin_port), p->branch, rport);
}
@@ -2543,9 +2755,21 @@
*/
static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod)
{
- struct sip_pkt *pkt;
+ struct sip_pkt *pkt = NULL;
int siptimer_a = DEFAULT_RETRANS;
int xmitres = 0;
+
+ /* If the transport is something reliable (TCP or TLS) then don't really send this reliably */
+ /* I removed the code from retrans_pkt that does the same thing so it doesn't get loaded into the scheduler */
+ /* According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited */
+ if (!(p->socket.type & SIP_TRANSPORT_UDP)) {
+ xmitres = __sip_xmit(dialog_ref(p), data, len); /* Send packet */
+ if (xmitres == XMIT_ERROR) { /* Serious network trouble, no need to try again */
+ append_history(p, "XmitErr", "%s", fatal ? "(Critical)" : "(Non-critical)");
+ return AST_FAILURE;
+ } else
+ return AST_SUCCESS;
+ }
if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
return AST_FAILURE;
@@ -2893,6 +3117,7 @@
ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp);
}
}
+
return tmp;
}
@@ -2929,7 +3154,7 @@
if (!strncasecmp(uri, scheme, l))
uri += l;
else {
- ast_log(LOG_NOTICE, "Missing scheme '%s' in '%s'\n", scheme, uri);
+ ast_debug(1, "Missing scheme '%s' in '%s'\n", scheme, uri);
error = -1;
}
}
@@ -3542,6 +3767,8 @@
*/
static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
{
+ dialog->socket = peer->socket;
+
if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) &&
(!peer->maxms || ((peer->lastms >= 0) && (peer->lastms <= peer->maxms)))) {
dialog->sa = (peer->addr.sin_addr.s_addr) ? peer->addr : peer->defaddr;
@@ -3701,13 +3928,13 @@
then hostname lookup */
hostn = peername;
- portno = port ? atoi(port) : STANDARD_SIP_PORT;
+ portno = port ? atoi(port) : (dialog->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
if (global_srvlookup) {
char service[MAXHOSTNAMELEN];
int tportno;
int ret;
- snprintf(service, sizeof(service), "_sip._udp.%s", peername);
+ snprintf(service, sizeof(service), "_sip._%s.%s", get_transport(dialog->socket.type), peername);
ret = ast_get_srv(NULL, host, sizeof(host), &tportno, service);
if (ret > 0) {
hostn = host;
@@ -3825,6 +4052,7 @@
p->t38.jointcapability = p->t38.capability;
ast_debug(2,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
+
xmitres = transmit_invite(p, SIP_INVITE, 1, 2);
if (xmitres == XMIT_ERROR)
return -1;
@@ -5284,6 +5512,8 @@
ast_mutex_init(&p->pvt_lock);
+ p->socket.fd = -1;
+ p->socket.type = SIP_TRANSPORT_UDP;
p->method = intended_method;
p->initid = -1;
p->waitid = -1;
@@ -5539,14 +5769,40 @@
{
struct sip_registry *reg;
int portnum = 0;
- char username[256] = "";
+ enum sip_transport transport = SIP_TRANSPORT_UDP;
+ char buf[256] = "";
+ char *username = NULL;
char *hostname=NULL, *secret=NULL, *authuser=NULL;
char *porta=NULL;
char *callback=NULL;
+ char *trans=NULL;
if (!value)
return -1;
- ast_copy_string(username, value, sizeof(username));
+
+ ast_copy_string(buf, value, sizeof(buf));
+
+ username = strstr(buf, "://");
+
+ if (username) {
+ *username = '\0';
+ username += 3;
+
+ trans = buf;
+
+ if (!strcasecmp(trans, "udp"))
+ transport = SIP_TRANSPORT_UDP;
+ else if (!strcasecmp(trans, "tcp"))
+ transport = SIP_TRANSPORT_TCP;
+ else if (!strcasecmp(trans, "tls"))
+ transport = SIP_TRANSPORT_TLS;
+ else
+ ast_log(LOG_WARNING, "'%s' is not a valid transport value for registration '%s' at line '%d'\n", trans, value, lineno);
+ } else {
+ username = buf;
+ ast_log(LOG_DEBUG, "no trans\n");
+ }
+
/* First split around the last '@' then parse the two components. */
hostname = strrchr(username, '@'); /* allow @ in the first part */
if (hostname)
@@ -5600,6 +5856,7 @@
ast_string_field_set(reg, authuser, authuser);
if (secret)
ast_string_field_set(reg, secret, secret);
+ reg->transport = transport;
reg->expire = -1;
reg->expiry = default_expiry;
reg->timeout = -1;
@@ -7045,6 +7302,12 @@
build_via(p);
ast_string_field_set(p, callid, callid);
+ p->socket.lock = req->socket.lock;
+ p->socket.type = req->socket.type;
+ p->socket.fd = req->socket.fd;
+ p->socket.port = req->socket.port;
+ p->socket.ser = req->socket.ser;
+
/* Use this temporary pvt structure to send the message */
__transmit_response(p, msg, req, XMIT_UNRELIABLE);
@@ -7933,10 +8196,13 @@
static void build_contact(struct sip_pvt *p)
{
/* Construct Contact: header */
- if (ntohs(p->ourip.sin_port) != STANDARD_SIP_PORT)
- ast_string_field_build(p, our_contact, "<sip:%s%s%s:%d>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr), ntohs(p->ourip.sin_port));
- else
- ast_string_field_build(p, our_contact, "<sip:%s%s%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr));
+ if (p->socket.type & SIP_TRANSPORT_UDP) {
+ if (!sip_standard_port(p->socket))
+ ast_string_field_build(p, our_contact, "<sip:%s%s%s:%d>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr), ntohs(p->socket.port));
+ else
+ ast_string_field_build(p, our_contact, "<sip:%s%s%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr));
+ } else
+ ast_string_field_build(p, our_contact, "<sip:%s%s%s:%d;transport=%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr), ntohs(p->socket.port), get_transport(p->socket.type));
}
/*! \brief Build the Remote Party-ID & From using callingpres options */
@@ -8085,8 +8351,8 @@
l = tmp_l;
}
- if (ntohs(p->ourip.sin_port) != STANDARD_SIP_PORT && ast_strlen_zero(p->fromdomain))
- snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", n, l, S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr)), ntohs(p->ourip.sin_port), p->tag);
+ if (!sip_standard_port(p->socket) && ast_strlen_zero(p->fromdomain))
+ snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", n, l, S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr)), ntohs(p->socket.port), p->tag);
else
snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=%s", n, l, S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr)), p->tag);
@@ -8367,7 +8633,7 @@
ast_copy_string(from, get_header(&p->initreq, "From"), sizeof(from));
c = get_in_brackets(from);
- if (strncasecmp(c, "sip:", 4)) {
+ if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c);
return -1;
}
@@ -8376,7 +8642,7 @@
ast_copy_string(to, get_header(&p->initreq, "To"), sizeof(to));
c = get_in_brackets(to);
- if (strncasecmp(c, "sip:", 4)) {
+ if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c);
return -1;
}
@@ -8682,6 +8948,7 @@
ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n");
return 0;
}
+
if (p->do_history)
append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
@@ -8733,6 +9000,9 @@
if (!ast_strlen_zero(r->callback))
ast_string_field_set(p, exten, r->callback);
+ /* Set transport and port so the correct contact is built */
+ p->socket.type = r->transport;
+ p->socket.port = htons(r->portno);
/*
check which address we should use in our contact header
based on whether the remote host is on the external or
@@ -8838,6 +9108,7 @@
r->regstate = auth ? REG_STATE_AUTHSENT : REG_STATE_REGSENT;
r->regattempts++; /* Another attempt */
ast_debug(4, "REGISTER attempt %d to %s@%s\n", r->regattempts, r->username, r->hostname);
+
return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
}
@@ -8892,10 +9163,12 @@
ast_copy_string(from, of, sizeof(from));
of = get_in_brackets(from);
ast_string_field_set(p, from, of);
- if (strncasecmp(of, "sip:", 4))
- ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
+ if (!strncasecmp(of, "sip:", 4))
+ of += 4;
+ else if (!strncasecmp(of, "sips:", 5))
+ of += 5;
else
- of += 4;
+ ast_log(LOG_NOTICE, "From address missing 'sip(s):', using it anyway\n");
/* Get just the username part */
if ((c = strchr(dest, '@')))
c = NULL;
@@ -9148,7 +9421,7 @@
ast_string_field_set(pvt, okcontacturi, c);
/* We should return false for URI:s we can't handle,
- like sips:, tel:, mailto:,ldap: etc */
+ like tel:, mailto:,ldap: etc */
return TRUE;
}
@@ -9159,7 +9432,7 @@
struct ast_hostent ahp;
int port;
char *host, *pt;
- char *contact;
+ char *contact, *contact2;
if (ast_test_flag(&pvt->flags[0], SIP_NAT_ROUTE)) {
@@ -9172,11 +9445,21 @@
/* Work on a copy */
contact = ast_strdupa(pvt->fullcontact);
+ contact2 = ast_strdupa(pvt->fullcontact);
/* We have only the part in <brackets> here so we just need to parse a SIP URI.*/
- if (parse_uri(contact, "sip:", &contact, NULL, &host, &pt, NULL))
- ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
- port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
+ if (pvt->socket.type == SIP_TRANSPORT_TLS) {
+ if (parse_uri(contact, "sips:", &contact, NULL, &host, &pt, NULL)) {
+ if (parse_uri(contact2, "sip:", &contact, NULL, &host, &pt, NULL))
+ ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
+ }
+ port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
+ } else {
+ if (parse_uri(contact, "sip:", &contact, NULL, &host, &pt, NULL))
+ ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
+ port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
+ }
+
ast_verbose("--- set_address_from_contact host '%s'\n", host);
/* XXX This could block for a long time XXX */
@@ -9201,7 +9484,7 @@
char data[BUFSIZ];
const char *expires = get_header(req, "Expires");
int expiry = atoi(expires);
- char *curi, *host, *pt;
+ char *curi, *host, *pt, *curi2;
int port;
const char *useragent;
struct hostent *hp;
@@ -9222,11 +9505,14 @@
}
}
+ pvt->socket = peer->socket = req->socket;
+
/* Look for brackets */
curi = contact;
if (strchr(contact, '<') == NULL) /* No <, check for ; and strip it */
strsep(&curi, ";"); /* This is Header options, not URI options */
curi = get_in_brackets(contact);
+ curi2 = ast_strdupa(curi);
/* if they did not specify Contact: or Expires:, they are querying
what we currently have stored as their contact address, so return
@@ -9264,9 +9550,18 @@
ast_string_field_build(pvt, our_contact, "<%s>", curi);
/* Make sure it's a SIP URL */
- if (parse_uri(curi, "sip:", &curi, NULL, &host, &pt, NULL))
- ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
- port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
+ if (pvt->socket.type == SIP_TRANSPORT_TLS) {
+ if (parse_uri(curi, "sips:", &curi, NULL, &host, &pt, NULL)) {
+ if (parse_uri(curi2, "sip:", &curi, NULL, &host, &pt, NULL))
+ ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
+ }
+ port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
+ } else {
+ if (parse_uri(curi, "sip:", &curi, NULL, &host, &pt, NULL))
+ ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
+ port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
+ }
+
oldsin = peer->addr;
if (!ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE)) {
/* XXX This could block for a long time XXX */
@@ -9302,7 +9597,8 @@
ast_sched_add(sched, (expiry + 10) * 1000, expire_register, peer);
pvt->expiry = expiry;
snprintf(data, sizeof(data), "%s:%d:%d:%s:%s", ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port), expiry, peer->username, peer->fullcontact);
- if (!peer->rt_fromcontact)
+ /* Saving TCP connections is useless, we won't be able to reconnect */
+ if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP))
ast_db_put("SIP/Registry", peer->name, data);
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\n", peer->name);
@@ -9738,6 +10034,8 @@
if (!strncasecmp(c, "sip:", 4)) {
name = c + 4;
+ } else if (!strncasecmp(c, "sips:", 5)) {
+ name = c + 5;
} else {
name = c;
ast_log(LOG_NOTICE, "Invalid to address: '%s' from %s (missing sip:) trying to use anyway...\n", c, ast_inet_ntoa(sin->sin_addr));
@@ -9775,6 +10073,7 @@
} else
res = AUTH_NOT_FOUND;
}
+
if (peer) {
/* Set Frame packetization */
if (p->rtp) {
@@ -9928,11 +10227,14 @@
return 0;
exten = get_in_brackets(tmp);
- if (strncasecmp(exten, "sip:", 4)) {
+ if (!strncasecmp(exten, "sip:", 4)) {
+ exten += 4;
+ } else if (!strncasecmp(exten, "sips:", 5)) {
+ exten += 5;
+ } else {
ast_log(LOG_WARNING, "Huh? Not an RDNIS SIP header (%s)?\n", exten);
return -1;
}
- exten += 4;
/* Get diversion-reason param if present */
if ((params = strchr(tmp, ';'))) {
@@ -9996,12 +10298,15 @@
ast_uri_decode(tmp);
uri = get_in_brackets(tmp);
-
- if (strncasecmp(uri, "sip:", 4)) {
+
+ if (!strncasecmp(uri, "sip:", 4)) {
+ uri += 4;
+ } else if (!strncasecmp(uri, "sips:", 5)) {
+ uri += 5;
+ } else {
ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", uri);
return -1;
}
- uri += 4;
/* Now find the From: caller ID and name */
/* XXX Why is this done in get_destination? Isn't it already done?
@@ -10015,11 +10320,14 @@
}
if (!ast_strlen_zero(from)) {
- if (strncasecmp(from, "sip:", 4)) {
+ if (!strncasecmp(from, "sip:", 4)) {
+ from += 4;
+ } else if (!strncasecmp(from, "sips:", 5)) {
+ from += 5;
+ } else {
ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", from);
return -1;
}
- from += 4;
if ((a = strchr(from, '@')))
*a++ = '\0';
else
@@ -10188,11 +10496,14 @@
if (pedanticsipchecking)
ast_uri_decode(refer_to);
- if (strncasecmp(refer_to, "sip:", 4)) {
+ if (!strncasecmp(refer_to, "sip:", 4)) {
+ refer_to += 4; /* Skip sip: */
+ } else if (!strncasecmp(refer_to, "sips:", 5)) {
+ refer_to += 5;
+ } else {
ast_log(LOG_WARNING, "Can't transfer to non-sip: URI. (Refer-to: %s)?\n", refer_to);
return -3;
}
- refer_to += 4; /* Skip sip: */
/* Get referred by header if it exists */
p_referred_by = get_header(req, "Referred-By");
@@ -10219,11 +10530,13 @@
}
referred_by_uri = get_in_brackets(h_referred_by);
- if(strncasecmp(referred_by_uri, "sip:", 4)) {
+ if (!strncasecmp(referred_by_uri, "sip:", 4)) {
+ referred_by_uri += 4; /* Skip sip: */
+ } else if (!strncasecmp(referred_by_uri, "sips:", 5)) {
+ referred_by_uri += 5; /* Skip sips: */
+ } else {
ast_log(LOG_WARNING, "Huh? Not a sip: header (Referred-by: %s). Skipping.\n", referred_by_uri);
referred_by_uri = NULL;
- } else {
- referred_by_uri += 4; /* Skip sip: */
}
}
@@ -10348,12 +10661,16 @@
if (pedanticsipchecking)
ast_uri_decode(c);
-
- if (strncasecmp(c, "sip:", 4)) {
+
+ if (!strncasecmp(c, "sip:", 4)) {
+ c += 4;
+ } else if (!strncasecmp(c, "sips:", 5)) {
+ c += 5;
+ } else {
ast_log(LOG_WARNING, "Huh? Not a SIP header in Also: transfer (%s)?\n", c);
return -1;
}
- c += 4;
+
if ((a = strchr(c, ';'))) /* Remove arguments */
*a = '\0';
@@ -10457,7 +10774,7 @@
if (c) {
*c = '\0';
c = ast_skip_blanks(c+1);
- if (strcasecmp(via, "SIP/2.0/UDP")) {
+ if (strcasecmp(via, "SIP/2.0/UDP") && strcasecmp(via, "SIP/2.0/TCP") && strcasecmp(via, "SIP/2.0/TLS")) {
ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via);
return;
}
@@ -10840,7 +11157,7 @@
char from[256];
char *dummy; /* dummy return value for parse_uri */
char *domain; /* dummy return value for parse_uri */
- char *of;
+ char *of, *of2;
char rpid_num[50];
const char *rpid;
enum check_auth_result res;
@@ -10868,6 +11185,8 @@
char *t = uri2;
if (!strncasecmp(t, "sip:", 4))
t+= 4;
+ else if (!strncasecmp(t, "sips:", 5))
+ t += 5;
ast_string_field_set(p, exten, t);
t = strchr(p->exten, '@');
if (t)
@@ -10878,10 +11197,19 @@
/* save the URI part of the From header */
ast_string_field_set(p, from, of);
+ of2 = ast_strdupa(of);
+
/* ignore all fields but name */
- if (parse_uri(of, "sip:", &of, &dummy, &domain, &dummy, &dummy)) {
- ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
- }
+ if (p->socket.type == SIP_TRANSPORT_TLS) {
+ if (parse_uri(of, "sips:", &of, &dummy, &domain, &dummy, &dummy)) {
+ if (parse_uri(of2, "sip:", &of, &dummy, &domain, &dummy, &dummy))
+ ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
+ }
+ } else {
+ if (parse_uri(of, "sip:", &of, &dummy, &domain, &dummy, &dummy))
+ ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
+ }
+
if (ast_strlen_zero(of)) {
/* XXX note: the original code considered a missing @host
* as a username-only URI. The SIP RFC (19.1.1) says that
@@ -11026,7 +11354,7 @@
/*! \brief CLI Command to show calls within limits set by call_limit */
static char *sip_show_inuse(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
-#define FORMAT "%-25.25s %-15.15s %-15.15s \n"
+#define FORMAT "%-25.25s %-15.15s %-15.15s \n"
#define FORMAT2 "%-25.25s %-15.15s %-15.15s \n"
char ilimits[40];
char iused[40];
@@ -11082,6 +11410,7 @@
#undef FORMAT2
}
+
/*! \brief Convert transfer mode to text string */
static char *transfermode2str(enum transfermodes mode)
{
@@ -11179,6 +11508,43 @@
static const char *cli_yesno(int x)
{
return x ? "Yes" : "No";
+}
+
+/*! \brief Show active TCP connections */
+static char *sip_show_tcp(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ struct sip_threadinfo *th;
+
+#define FORMAT2 "%-30.30s %3.6s %9.9s %6.6s\n"
+#define FORMAT "%-30.30s %-6d %-9.9s %-6.6s\n"
+
+ switch (cmd) {
+ case CLI_INIT:
+ e->command = "sip show tcp";
+ e->usage =
+ "Usage: sip show tcp\n"
+ " Lists all active TCP/TLS sessions.\n";
+ return NULL;
+ case CLI_GENERATE:
+ return NULL;
+ }
+
+ if (a->argc != 3)
+ return CLI_SHOWUSAGE;
+
+ ast_cli(a->fd, FORMAT2, "Host", "Port", "Transport", "Type");
+ AST_LIST_LOCK(&threadl);
+ AST_LIST_TRAVERSE(&threadl, th, list) {
+ ast_cli(a->fd, FORMAT, ast_inet_ntoa(th->ser->requestor.sin_addr),
+ ntohs(th->ser->requestor.sin_port),
+ get_transport(th->type),
+ (th->ser->client ? "Client" : "Server"));
+
+ }
+ AST_LIST_UNLOCK(&threadl);
+ return CLI_SUCCESS;
+#undef FORMAT
+#undef FORMAT2
}
/*! \brief CLI Command 'SIP Show Users' */
@@ -11960,6 +12326,7 @@
ast_cli(fd, " ToHost : %s\n", peer->tohost);
ast_cli(fd, " Addr->IP : %s Port %d\n", peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port));
ast_cli(fd, " Defaddr->IP : %s Port %d\n", ast_inet_ntoa(peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
+ ast_cli(fd, " Transport : %s\n", get_transport(peer->socket.type));
if (!ast_strlen_zero(global_regcontext))
ast_cli(fd, " Reg. exten : %s\n", peer->regexten);
ast_cli(fd, " Def. Username: %s\n", peer->username);
@@ -13755,6 +14122,8 @@
if (ast_test_flag(&p->flags[0], SIP_PROMISCREDIR)) {
if (!strncasecmp(s, "sip:", 4))
s += 4;
+ else if (!strncasecmp(s, "sips:", 5))
+ s += 5;
e = strchr(s, '/');
if (e)
*e = '\0';
@@ -13776,6 +14145,8 @@
if (!strncasecmp(s, "sip:", 4))
s += 4;
+ else if (!strncasecmp(s, "sips:", 5))
+ s += 5;
e = strchr(s, ';'); /* And username ; parameters? */
if (e)
*e = '\0';
@@ -15106,7 +15477,7 @@
ast_softhangup_nolock(transferer->chan1, AST_SOFTHANGUP_DEV);
if (target->chan1)
ast_softhangup_nolock(target->chan1, AST_SOFTHANGUP_DEV);
- return -2;
+ return -1;
}
return 0;
}
@@ -16203,7 +16574,6 @@
/* Failed transfer */
transmit_notify_with_sipfrag(transferer, seqno, "486 Busy Here", TRUE);
append_history(transferer, "Xfer", "Refer failed");
- transferer->refer->status = REFER_FAILED;
if (targetcall_pvt->owner)
ast_channel_unlock(targetcall_pvt->owner);
/* Right now, we have to hangup, sorry. Bridge is destroyed */
@@ -17362,15 +17732,11 @@
{
struct sip_request req;
struct sockaddr_in sin = { 0, };
- struct sip_pvt *p;
int res;
socklen_t len = sizeof(sin);
- int nounlock;
- int recount = 0;
- int lockretry;
memset(&req, 0, sizeof(req));
- res = recvfrom(sipsock, req.data, sizeof(req.data) - 1, 0, (struct sockaddr *)&sin, &len);
+ res = recvfrom(fd, req.data, sizeof(req.data) - 1, 0, (struct sockaddr *)&sin, &len);
if (res < 0) {
#if !defined(__FreeBSD__)
if (errno == EAGAIN)
@@ -17387,20 +17753,42 @@
} else
req.data[res] = '\0';
req.len = res;
- if(sip_debug_test_addr(&sin)) /* Set the debug flag early on packet level */
- req.debug = 1;
+
+ req.socket.fd = sipsock;
+ req.socket.type = SIP_TRANSPORT_UDP;
+ req.socket.ser = NULL;
+ req.socket.port = htons(bindaddr.sin_port);
+ req.socket.lock = NULL;
+
+ handle_request_do(&req, &sin);
+
+ return 1;
+}
+
+static int handle_request_do(struct sip_request *req, struct sockaddr_in *sin)
+{
+ struct sip_pvt *p;
+ int recount = 0;
+ int nounlock = 0;
+ int lockretry;
+
+ 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 (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 (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 */
+ req->len = lws2sws(req->data, req->len); /* Fix multiline headers */
+ if (req->debug) {
+ ast_verbose("\n<--- SIP read from %s://%s:%d --->\n%s\n<------------->\n",
+ get_transport(req->socket.type), ast_inet_ntoa(sin->sin_addr),
+ ntohs(sin->sin_port), req->data);
+ }
+
+ parse_request(req);
+ req->method = find_sip_method(req->rlPart1);
+
+ 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 */
return 1;
/* Process request, with netlock held, and with usual deadlock avoidance */
@@ -17408,12 +17796,15 @@
ast_mutex_lock(&netlock);
/* Find the active SIP dialog or create a new one */
- p = find_call(&req, &sin, req.method); /* returns p locked */
+ p = find_call(req, sin, req->method); /* returns p locked */
if (p == NULL) {
- ast_debug(1, "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);
ast_mutex_unlock(&netlock);
return 1;
}
+
+ p->socket = req->socket;
+
/* Go ahead and lock the owner if it has one -- we may need it */
/* becaues this is deadlock-prone, we need to try and unlock if failed */
if (!p->owner || !ast_channel_trylock(p->owner))
@@ -17424,35 +17815,117 @@
/* Sleep for a very short amount of time */
usleep(1);
}
- p->recv = sin;
+ p->recv = *sin;
if (p->do_history) /* This is a request or response, note what it was for */
- append_history(p, "Rx", "%s / %s / %s", req.data, get_header(&req, "CSeq"), req.rlPart2);
+ append_history(p, "Rx", "%s / %s / %s", req->data, get_header(req, "CSeq"), req->rlPart2);
if (!lockretry) {
if (p->owner)
ast_log(LOG_ERROR, "We could NOT get the channel lock for %s! \n", S_OR(p->owner->name, "- no channel name ??? - "));
ast_log(LOG_ERROR, "SIP transaction failed: %s \n", p->callid);
- if (req.method != SIP_ACK)
- transmit_response(p, "503 Server error", &req); /* We must respond according to RFC 3261 sec 12.2 */
+ if (req->method != SIP_ACK)
+ transmit_response(p, "503 Server error", req); /* We must respond according to RFC 3261 sec 12.2 */
/* XXX We could add retry-after to make sure they come back */
append_history(p, "LockFail", "Owner lock failed, transaction failed.");
return 1;
}
+
nounlock = 0;
- if (handle_incoming(p, &req, &sin, &recount, &nounlock) == -1) {
+ if (handle_incoming(p, req, sin, &recount, &nounlock) == -1) {
/* Request failed */
ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
}
+ if (recount)
+ ast_update_use_count();
+
if (p->owner && !nounlock)
ast_channel_unlock(p->owner);
sip_pvt_unlock(p);
ast_mutex_unlock(&netlock);
- if (recount)
- ast_update_use_count();
return 1;
+}
+
[... 785 lines stripped ...]
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