[asterisk-commits] oej: branch oej/videocaps r128559 - in /team/oej/videocaps: ./ channels/ conf...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sun Jul 6 15:54:53 CDT 2008
Author: oej
Date: Sun Jul 6 15:54:52 2008
New Revision: 128559
URL: http://svn.digium.com/view/asterisk?view=rev&rev=128559
Log:
Reset automerge, resolve conflict.
(someone just killed the user... Weird)
Added:
team/oej/videocaps/doc/realtimetext.txt
- copied unchanged from r128491, trunk/doc/realtimetext.txt
Modified:
team/oej/videocaps/ (props changed)
team/oej/videocaps/channels/chan_sip.c
team/oej/videocaps/configs/sip.conf.sample
team/oej/videocaps/include/asterisk/channel.h
team/oej/videocaps/include/asterisk/pbx.h
team/oej/videocaps/main/cli.c
team/oej/videocaps/main/manager.c
team/oej/videocaps/main/pbx.c
team/oej/videocaps/pbx/pbx_realtime.c
team/oej/videocaps/res/res_agi.c
Propchange: team/oej/videocaps/
------------------------------------------------------------------------------
automerge = http://www.codename-pineapple.org/
Propchange: team/oej/videocaps/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Jul 6 15:54:52 2008
@@ -1,1 +1,1 @@
-/trunk:1-128220
+/trunk:1-128512
Modified: team/oej/videocaps/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/channels/chan_sip.c?view=diff&rev=128559&r1=128558&r2=128559
==============================================================================
--- team/oej/videocaps/channels/chan_sip.c (original)
+++ team/oej/videocaps/channels/chan_sip.c Sun Jul 6 15:54:52 2008
@@ -40,6 +40,8 @@
* \todo We need to test TCP sessions with SIP proxies and in regards
* to the SIP outbound specs.
* \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
+ * \todo Save TCP/TLS sessions in registry
+ * \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
*
* \ingroup channel_drivers
*
@@ -240,7 +242,6 @@
static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
-static int expiry = DEFAULT_EXPIRY;
#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
@@ -298,9 +299,10 @@
};
+/*! \brief The result of a lot of functions */
enum sip_result {
- AST_SUCCESS = 0,
- AST_FAILURE = -1,
+ AST_SUCCESS = 0, /*! FALSE means success, funny enough */
+ AST_FAILURE = -1,
};
/*! \brief States for the INVITE transaction, not the dialog
@@ -322,7 +324,7 @@
\note Should be aligned to above table as index */
static const struct invstate2stringtable {
const enum invitestates state;
- const char const *desc;
+ const char *desc;
} invitestate2string[] = {
{INV_NONE, "None" },
{INV_CALLING, "Calling (Trying)"},
@@ -334,6 +336,9 @@
{INV_CANCELLED, "Cancelled"}
};
+/*! \brief When sending a SIP message, we can send with a few options, depending on
+ type of SIP request. UNRELIABLE is moslty used for responses to repeated requests,
+ where the original response would be sent RELIABLE in an INVITE transaction */
enum xmittype {
XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
If it fails, it's critical and will cause a teardown of the session */
@@ -347,6 +352,7 @@
PARSE_REGISTER_QUERY,
};
+/*! \brief Type of subscription, based on the packages we do support */
enum subscriptiontype {
NONE = 0,
XPIDF_XML,
@@ -454,11 +460,13 @@
SESSION_TIMER_REFRESHER_UAS /*!< Session is refreshed by the UAS */
};
-/*!< Define some SIP transports */
+/*! \brief Define some implemented SIP transports
+ \note Asterisk does not support SCTP or UDP/DTLS
+*/
enum sip_transport {
- SIP_TRANSPORT_UDP = 1,
- SIP_TRANSPORT_TCP = 1 << 1,
- SIP_TRANSPORT_TLS = 1 << 2,
+ SIP_TRANSPORT_UDP = 1, /*!< Unreliable transport for SIP, needs retransmissions */
+ SIP_TRANSPORT_TCP = 1 << 1, /*!< Reliable, but unsecure */
+ SIP_TRANSPORT_TLS = 1 << 2, /*!< TCP/TLS - reliable and secure transport for signalling */
};
/*! \brief definition of a sip proxy server
@@ -645,12 +653,18 @@
*/
#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
-/*! \brief SIP Extensions we support */
+/*! \brief SIP Extensions we support
+ \note This should be generated based on the previous array
+ in combination with settings.
+ \todo We should not have "timer" if it's disabled in the configuration file.
+*/
#define SUPPORTED_EXTENSIONS "replaces, timer"
-/*! \brief Standard SIP and TLS port from RFC 3261. DO NOT CHANGE THIS */
+/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT 5060
+/*! \brief Standard SIP TLS port for sips: from RFC 3261. DO NOT CHANGE THIS */
#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,
@@ -681,12 +695,12 @@
#define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_TEXT 0 /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_COS_SIP 4
-#define DEFAULT_COS_AUDIO 5
-#define DEFAULT_COS_VIDEO 6
-#define DEFAULT_COS_TEXT 5
-#define DEFAULT_ALLOW_EXT_DOM TRUE
-#define DEFAULT_REALM "asterisk"
+#define DEFAULT_COS_SIP 4 /*!< Level 2 class of service for SIP signalling */
+#define DEFAULT_COS_AUDIO 5 /*!< Level 2 class of service for audio media */
+#define DEFAULT_COS_VIDEO 6 /*!< Level 2 class of service for video media */
+#define DEFAULT_COS_TEXT 5 /*!< Level 2 class of service for text media (T.140) */
+#define DEFAULT_ALLOW_EXT_DOM TRUE /*!< Allow external domains */
+#define DEFAULT_REALM "asterisk" /*!< Realm for HTTP digest authentication */
#define DEFAULT_NOTIFYRINGING TRUE
#define DEFAULT_PEDANTIC FALSE
#define DEFAULT_AUTOCREATEPEER FALSE
@@ -700,7 +714,7 @@
#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
-#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
+#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
#endif
/*@}*/
@@ -741,7 +755,6 @@
*/
/*@{*/
static int global_directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */
-static int global_limitonpeers; /*!< Match call limit on peers only */
static int global_rtautoclear; /*!< Realtime ?? */
static int global_notifyringing; /*!< Send notifications on ringing */
static int global_notifyhold; /*!< Send notifications on hold */
@@ -756,7 +769,7 @@
static int global_rtpkeepalive; /*!< Send RTP keepalives */
static int global_reg_timeout;
static int global_regattempts_max; /*!< Registration attempts before giving up */
-static int global_allowguest; /*!< allow unauthenticated users/peers to connect? */
+static int global_allowguest; /*!< allow unauthenticated peers to connect? */
static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer
call-limit to 999. When we remove the call-limit from the code, we can make it
with just a boolean flag in the device structure */
@@ -783,13 +796,13 @@
static int global_authfailureevents; /*!< Whether we send authentication failure manager events or not. Default no. */
static int global_t1; /*!< T1 time */
static int global_t1min; /*!< T1 roundtrip time minimum */
-static int global_timer_b; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
-static int global_regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
-static int global_autoframing; /*!< Turn autoframing on or off. */
+static int global_timer_b; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
+static int global_regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
+static int global_autoframing; /*!< Turn autoframing on or off. */
static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */
static struct sip_proxy global_outboundproxy; /*!< Outbound proxy */
-static int global_matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
-static int global_qualifyfreq; /*!< Qualify frequency */
+static int global_matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
+static int global_qualifyfreq; /*!< Qualify frequency */
static int global_videoupdate = DEFAULT_VIDEO_UPDATE;
/* We set default caps but make them invalid.
If a rtpmap comes in the we set valid.
@@ -800,6 +813,7 @@
/*! \brief Codecs that we support by default: */
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
+
static enum st_mode global_st_mode; /*!< Mode of operation for Session-Timers */
static enum st_refresher global_st_refresher; /*!< Session-Timer refresher */
static int global_min_se; /*!< Lowest threshold for session refresh interval */
@@ -810,8 +824,6 @@
/*! \name Object counters @{
* \bug These counters are not handled in a thread-safe way ast_atomic_fetchadd_int()
* should be used to modify these values. */
-static int suserobjs = 0; /*!< Static users */
-static int ruserobjs = 0; /*!< Realtime users */
static int speerobjs = 0; /*!< Static peers */
static int rpeerobjs = 0; /*!< Realtime peers */
static int apeerobjs = 0; /*!< Autocreated peer objects */
@@ -819,14 +831,13 @@
/* }@ */
static struct ast_flags global_flags[2] = {{0}}; /*!< global SIP_ flags */
-static char used_context[AST_MAX_CONTEXT]; /*!< name of automatically created context for unloading */
+static char used_context[AST_MAX_CONTEXT]; /*!< name of automatically created context for unloading */
AST_MUTEX_DEFINE_STATIC(netlock);
/*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
when it's doing something critical. */
-
AST_MUTEX_DEFINE_STATIC(monlock);
AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
@@ -847,12 +858,12 @@
#define DEC_CALL_RINGING 2
#define INC_CALL_RINGING 3
-/*!< The SIP socket definition */
+/*! \brief The SIP socket definition */
struct sip_socket {
- enum sip_transport type;
- int fd;
+ enum sip_transport type; /*!< UDP, TCP or TLS */
+ int fd; /*!< Filed descriptor, the actual socket */
uint16_t port;
- struct ast_tcptls_session_instance *ser;
+ struct ast_tcptls_session_instance *ser; /* If tcp or tls, a socket manager */
};
/*! \brief sip_request: The data grabbed from the UDP socket
@@ -1121,8 +1132,8 @@
static enum sip_debug_e sipdebug;
/*! \brief extra debugging for 'text' related events.
- * At thie moment this is set together with sip_debug_console.
- * It should either go away or be implemented properly.
+ * At the moment this is set together with sip_debug_console.
+ * \note It should either go away or be implemented properly.
*/
static int sipdebug_text;
@@ -1234,7 +1245,7 @@
-/*! \brief sip_pvt: structures used for each SIP dialog, ie. a call, a registration, a subscribe.
+/*! \brief Structure used for each SIP dialog, ie. a call, a registration, a subscribe.
* Created and initialized by sip_alloc(), the descriptor goes into the list of
* descriptors (dialoglist).
*/
@@ -1391,7 +1402,7 @@
/*! Max entires in the history list for a sip_pvt */
#define MAX_HISTORY_ENTRIES 50
-/*!
+/*! \brief
* Here we implement the container for dialogs (sip_pvt), defining
* generic wrapper functions to ease the transition from the current
* implementation (a single linked list) to a different container.
@@ -1401,7 +1412,11 @@
*/
struct ao2_container *dialogs;
-/*!
+#define sip_pvt_lock(x) ao2_lock(x)
+#define sip_pvt_trylock(x) ao2_trylock(x)
+#define sip_pvt_unlock(x) ao2_unlock(x)
+
+/*! \brief
* when we create or delete references, make sure to use these
* functions so we keep track of the refcounts.
* To simplify the code, we allow a NULL to be passed to dialog_unref().
@@ -1409,6 +1424,7 @@
#ifdef REF_DEBUG
#define dialog_ref(arg1,arg2) dialog_ref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
#define dialog_unref(arg1,arg2) dialog_unref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
+
static struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
{
if (p)
@@ -1462,46 +1478,6 @@
int packetlen; /*!< Length of packet */
struct ast_str *data;
};
-
-/*! \brief Structure for SIP user data. User's place calls to us */
-struct sip_user {
- /* Users who can access various contexts */
- char name[80];
- char secret[80]; /*!< Password */
- char md5secret[80]; /*!< Password in md5 */
- char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
- char subscribecontext[AST_MAX_CONTEXT]; /* Default context for subscriptions */
- char cid_num[80]; /*!< Caller ID num */
- char cid_name[80]; /*!< Caller ID name */
- char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */
- char language[MAX_LANGUAGE]; /*!< Default language for this user */
- char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
- char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
- char parkinglot[AST_MAX_CONTEXT];/*!< Parkinglot */
- char useragent[256]; /*!< User agent in SIP request */
- struct ast_codec_pref prefs; /*!< codec prefs */
- ast_group_t callgroup; /*!< Call group */
- ast_group_t pickupgroup; /*!< Pickup Group */
- unsigned int sipoptions; /*!< Supported SIP options */
- struct ast_flags flags[2]; /*!< SIP_ flags */
-
- /* things that don't belong in flags */
- char is_realtime; /*!< this is a 'realtime' user */
- unsigned int the_mark:1; /*!< moved out of the ASTOBJ fields; that which bears the_mark should be deleted! */
-
- int amaflags; /*!< AMA flags for billing */
- int callingpres; /*!< Calling id presentation */
- struct ast_capabilities caps; /*!< User capability */
- int capability; /*!< Codec capability */
- int inUse; /*!< Number of calls in use */
- int call_limit; /*!< Limit of concurrent calls */
- enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
- struct ast_ha *ha; /*!< ACL setting */
- struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
- int videoupdate /*!< Defines use of XML or RTCP for video update */;
- int autoframing;
- struct sip_st_cfg stimer; /*!< SIP Session-Timers */
-};
/*!
* \brief A peer's mailbox
@@ -1564,7 +1540,8 @@
char rt_fromcontact; /*!< P: copy fromcontact from realtime */
char host_dynamic; /*!< P: Dynamic Peers register with Asterisk */
char selfdestruct; /*!< P: Automatic peers need to destruct themselves */
- char the_mark; /*!< moved out of ASTOBJ into struct proper; That which bears the_mark should be deleted! */
+ char onlymatchonip; /*!< P: Only match on IP for incoming calls (old type=peer) */
+ char the_mark; /*!< moved out of ASTOBJ into struct proper; That which bears the_mark should be deleted! */
int expire; /*!< When to expire this peer registration */
int capability; /*!< Codec capability */
@@ -1630,7 +1607,7 @@
AST_STRING_FIELD(callback); /*!< Contact extension */
AST_STRING_FIELD(random);
);
- enum sip_transport transport;
+ enum sip_transport transport; /*!< Transport for this registration UDP, TCP or TLS */
int portno; /*!< Optional port override */
int expire; /*!< Sched ID of expiration */
int expiry; /*!< Value to use for the Expires header */
@@ -1648,11 +1625,12 @@
char lastmsg[256]; /*!< Last Message sent/received */
};
+/*! \brief Definition of a thread that handles a socket */
struct sip_threadinfo {
int stop;
pthread_t threadid;
struct ast_tcptls_session_instance *ser;
- enum sip_transport type; /* We keep a copy of the type here so we can display it in the connection list */
+ 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;
};
@@ -1663,7 +1641,7 @@
static int hash_dialog_size = 17;
static int hash_user_size = 17;
#else
-static int hash_peer_size = 563;
+static int hash_peer_size = 563; /*!< Size of peer hash table, prime number preferred! */
static int hash_dialog_size = 563;
static int hash_user_size = 563;
#endif
@@ -1671,10 +1649,7 @@
/*! \brief The thread list of TCP threads */
static AST_LIST_HEAD_STATIC(threadl, sip_threadinfo);
-/*! \brief The user list: Users and friends */
-static struct ao2_container *users;
-
-/*! \brief The peer list: Peers and Friends */
+/*! \brief The peer list: Users, Peers and Friends */
struct ao2_container *peers;
struct ao2_container *peers_by_ip;
@@ -1684,7 +1659,7 @@
int recheck;
} regl;
-/*!
+/*! \brief
* \note The only member of the peer used here is the name field
*/
static int peer_hash_cb(const void *obj, const int flags)
@@ -1738,26 +1713,6 @@
return 0;
}
return CMP_MATCH;
-}
-
-/*!
- * \note The only member of the user used here is the name field
- */
-static int user_hash_cb(const void *obj, const int flags)
-{
- const struct sip_user *user = obj;
-
- return ast_str_hash(user->name);
-}
-
-/*!
- * \note The only member of the user used here is the name field
- */
-static int user_cmp_cb(void *obj, void *arg, int flags)
-{
- struct sip_user *user = obj, *user2 = arg;
-
- return !strcasecmp(user->name, user2->name) ? CMP_MATCH : 0;
}
/*!
@@ -1817,7 +1772,7 @@
*/
static int sipsock = -1;
-static struct sockaddr_in bindaddr; /*!< The address we bind to */
+static struct sockaddr_in bindaddr; /*!< UDP: The address we bind to */
/*! \brief our (internal) default address/port to put in SIP/SDP messages
* internip is initialized picking a suitable address from one of the
@@ -1860,8 +1815,8 @@
*/
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 int ourport_tcp; /*!< The port used for TCP connections */
+static int ourport_tls; /*!< The port used for TCP/TLS connections */
static struct sockaddr_in debugaddr;
static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
@@ -2043,12 +1998,9 @@
static char *transfermode2str(enum transfermodes mode) attribute_const;
static const char *nat2str(int nat) attribute_const;
static int peer_status(struct sip_peer *peer, char *status, int statuslen);
-static char *sip_show_users(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_sched(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char * _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_show_peers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *_sip_dbdump(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]);
-static char *sip_dbdump(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_objects(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static void print_group(int fd, ast_group_t group, int crlf);
static const char *dtmfmode2str(int mode) attribute_const;
@@ -2060,7 +2012,6 @@
static char *sip_show_domains(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_show_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static char *sip_show_user(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *_sip_qualify_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_qualify_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_registry(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
@@ -2073,8 +2024,6 @@
static char *complete_sip_show_history(const char *line, const char *word, int pos, int state);
static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
static char *complete_sip_unregister(const char *line, const char *word, int pos, int state);
-static char *complete_sip_user(const char *word, int state, int flags2);
-static char *complete_sip_show_user(const char *line, const char *word, int pos, int state);
static char *complete_sipnotify(const char *line, const char *word, int pos, int state);
static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
@@ -2108,18 +2057,14 @@
/*--- Device object handling */
static struct sip_peer *temp_peer(const char *name);
-static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime);
-static struct sip_user *build_user(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime);
+static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime, int ispeer);
static int update_call_counter(struct sip_pvt *fup, int event);
static void sip_destroy_peer(struct sip_peer *peer);
static void sip_destroy_peer_fn(void *peer);
-static void sip_destroy_user(struct sip_user *user);
-static void sip_destroy_user_fn(void *user);
static void set_peer_defaults(struct sip_peer *peer);
static struct sip_peer *temp_peer(const char *name);
static void register_peer_exten(struct sip_peer *peer, int onoff);
-static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime);
-static struct sip_user *find_user(const char *name, int realtime);
+static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime, int forcenamematch);
static int sip_poke_peer_s(const void *data);
static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
static void reg_source_db(struct sip_peer *peer);
@@ -2129,7 +2074,6 @@
/* Realtime device support */
static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, const char *useragent, int expirey, int deprecated_username);
-static struct sip_user *realtime_user(const char *username);
static void update_peer(struct sip_peer *p, int expiry);
static struct ast_variable *get_insecure_variable_from_config(struct ast_config *config);
static const char *get_name_from_variable(struct ast_variable *var, const char *newpeername);
@@ -2176,6 +2120,10 @@
static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
static int get_msg_text(char *buf, int len, struct sip_request *req, int addnewline);
static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout);
+
+/*-- TCP connection handling ---*/
+static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *ser);
+static void *sip_tcp_worker_fn(void *);
/*--- Constructing requests and responses */
static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
@@ -2298,11 +2246,14 @@
*/
static struct ast_channel_tech sip_tech_info;
-static void *sip_tcp_worker_fn(void *);
-
+
+/*! \brief Working TLS connection configuration */
static struct ast_tls_config sip_tls_cfg;
+
+/*! \brief Default TLS connection configuration */
static struct ast_tls_config default_tls_cfg;
+/*! \brief The TCP server definition */
static struct server_args sip_tcp_desc = {
.accept_fd = -1,
.master = AST_PTHREADT_NULL,
@@ -2313,6 +2264,7 @@
.worker_fn = sip_tcp_worker_fn,
};
+/*! \brief The TCP/TLS server definition */
static struct server_args sip_tls_desc = {
.accept_fd = -1,
.master = AST_PTHREADT_NULL,
@@ -2363,8 +2315,8 @@
.get_codec = sip_get_codec,
};
-static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *ser);
-
+
+/*! \brief SIP TCP connection handler */
static void *sip_tcp_worker_fn(void *data)
{
struct ast_tcptls_session_instance *ser = data;
@@ -2372,7 +2324,7 @@
return _sip_tcp_helper_thread(NULL, ser);
}
-/*! \brief SIP TCP helper function */
+/*! \brief SIP TCP thread management function */
static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *ser)
{
int res, cl;
@@ -2465,8 +2417,10 @@
fclose(ser->f);
ser->f = NULL;
ser->fd = -1;
- if (reqcpy.data)
+ if (reqcpy.data) {
ast_free(reqcpy.data);
+ }
+
if (req.data) {
ast_free(req.data);
req.data = NULL;
@@ -2479,9 +2433,6 @@
return NULL;
}
-#define sip_pvt_lock(x) ao2_lock(x)
-#define sip_pvt_trylock(x) ao2_trylock(x)
-#define sip_pvt_unlock(x) ao2_unlock(x)
/*!
* helper functions to unreference various types of objects.
@@ -2491,12 +2442,6 @@
static void *unref_peer(struct sip_peer *peer, char *tag)
{
ao2_t_ref(peer, -1, tag);
- return NULL;
-}
-
-static void *unref_user(struct sip_user *user, char *tag)
-{
- ao2_t_ref(user, -1, tag);
return NULL;
}
@@ -4060,7 +4005,7 @@
/* Peer found in realtime, now build it in memory */
- peer = build_peer(newpeername, var, varregs, 1);
+ peer = build_peer(newpeername, var, varregs, TRUE, FALSE);
if (!peer) {
if(peerlist)
ast_config_destroy(peerlist);
@@ -4099,13 +4044,14 @@
return peer;
}
-/*! \brief Locate peer by name or ip address
- * This is used on incoming SIP message to find matching peer on ip
- or outgoing message to find matching peer on name
+/*! \brief Locate device by name or ip address
+ * This is used on find matching device on name or ip/port.
+ If the device was declared as type=peer, we don't match on peer name on incoming INVITEs.
+
\note Avoid using this function in new functions if there is a way to avoid it, i
since it might cause a database lookup.
*/
-static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime)
+static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime, int forcenamematch)
{
struct sip_peer *p = NULL;
struct sip_peer tmp_peer;
@@ -4131,94 +4077,6 @@
p = realtime_peer(peer, sin);
return p;
-}
-
-static void sip_destroy_user_fn(void *user)
-{
- sip_destroy_user(user);
-}
-
-
-/*! \brief Remove user object from in-memory storage */
-static void sip_destroy_user(struct sip_user *user)
-{
- ast_debug(3, "Destroying user object from memory: %s\n", user->name);
-
- ast_free_ha(user->ha);
- if (user->chanvars) {
- ast_variables_destroy(user->chanvars);
- user->chanvars = NULL;
- }
- if (user->is_realtime)
- ast_atomic_fetchadd_int(&ruserobjs, -1);
- else
- ast_atomic_fetchadd_int(&suserobjs, -1);
-}
-
-/*! \brief Load user from realtime storage
- * Loads user from "sipusers" category in realtime (extconfig.conf)
- * returns a refcounted pointer to a sip_user structure.
- * Users are matched on From: user name (the domain in skipped) */
-static struct sip_user *realtime_user(const char *username)
-{
- struct ast_variable *var;
- struct ast_variable *tmp;
- struct sip_user *user = NULL;
-
- var = ast_load_realtime("sipusers", "name", username, SENTINEL);
-
- if (!var)
- return NULL;
-
- for (tmp = var; tmp; tmp = tmp->next) {
- if (!strcasecmp(tmp->name, "type") &&
- !strcasecmp(tmp->value, "peer")) {
- ast_variables_destroy(var);
- return NULL;
- }
- }
-
- user = build_user(username, var, NULL, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
-
- if (!user) { /* No user found */
- ast_variables_destroy(var);
- return NULL;
- }
-
- if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
- ast_set_flag(&user->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
- ast_atomic_fetchadd_int(&suserobjs, 1);
- ao2_t_link(users, user, "link user into users table");
- } else {
- /* Move counter from s to r... */
- ast_atomic_fetchadd_int(&suserobjs, -1);
- ast_atomic_fetchadd_int(&ruserobjs, 1);
- user->is_realtime = 1;
- }
- ast_variables_destroy(var);
- return user;
-}
-
-/*! \brief Locate user by name
- * Locates user by name (From: sip uri user name part) first
- * from in-memory list (static configuration) then from
- * realtime storage (defined in extconfig.conf)
- * \return a refcounted pointer to a user. Make sure the
- * the ref count is decremented when this pointer is nulled, changed,
- * or goes out of scope!
- */
-
-static struct sip_user *find_user(const char *name, int realtime)
-{
- struct sip_user tmp;
- struct sip_user *u;
-
- ast_copy_string(tmp.name, name, sizeof(tmp.name));
- u = ao2_t_find(users, &tmp, OBJ_POINTER, "ao2_find in users table");
-
- if (!u && realtime)
- u = realtime_user(name);
- return u;
}
/*! \brief Set nat mode on the various data sockets */
@@ -4438,7 +4296,7 @@
return 0;
}
-/*! \brief create address structure from peer name
+/*! \brief create address structure from device name
* Or, if peer not found, find it in the global DNS
* returns TRUE (-1) on failure, FALSE on success */
static int create_addr(struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin)
@@ -4459,7 +4317,7 @@
dialog->sa.sin_family = AF_INET;
dialog->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
dialog->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
- peer = find_peer(peername, NULL, 1);
+ peer = find_peer(peername, NULL, TRUE, TRUE);
if (peer) {
int res = create_addr_from_peer(dialog, peer);
@@ -4492,6 +4350,9 @@
/* Let's see if we can find the host in DNS. First try DNS SRV records,
then hostname lookup */
+ /*! \todo Fix this function. When we ask SRC, we should check all transports
+ In the future, we should first check NAPTR to find out transport preference
+ */
hostn = peername;
portno = port ? atoi(port) : (dialog->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
if (global_srvlookup) {
@@ -4776,7 +4637,7 @@
}
}
-/*! \brief update_call_counter: Handle call_limit for SIP users
+/*! \brief update_call_counter: Handle call_limit for SIP devices
* Setting a call-limit will cause calls above the limit not to be accepted.
*
* Remember that for a type=friend, there's one limit for the user and
@@ -4795,7 +4656,6 @@
char name[256];
int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
int outgoing = fup->outgoing_call;
- struct sip_user *u = NULL;
struct sip_peer *p = NULL;
ast_debug(3, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
@@ -4808,18 +4668,14 @@
ast_copy_string(name, fup->username, sizeof(name));
- /* Check the list of users only for incoming calls */
- if (global_limitonpeers == FALSE && !outgoing && (u = find_user(name, 1))) {
- inuse = &u->inUse;
- call_limit = &u->call_limit;
- inringing = NULL;
- } else if ( (p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, 1) ) ) { /* Try to find peer */
+ /* Check the list of devices */
+ if ( (p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, TRUE, FALSE) ) ) { /* Try to find peer */
inuse = &p->inUse;
call_limit = &p->call_limit;
inringing = &p->inRinging;
ast_copy_string(name, fup->peername, sizeof(name));
}
- if (!p && !u) {
+ if (!p) {
ast_debug(2, "%s is not a local device, no call limit\n", name);
return 0;
}
@@ -4844,7 +4700,7 @@
sip_peer_hold(fup, FALSE);
}
if (sipdebug)
- ast_debug(2, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
+ ast_debug(2, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", "peer", name, *call_limit);
break;
case INC_CALL_RINGING:
@@ -4852,11 +4708,8 @@
/* If call limit is active and we have reached the limit, reject the call */
if (*call_limit > 0 ) {
if (*inuse >= *call_limit) {
- ast_log(LOG_WARNING, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
- if (u)
- unref_user(u, "update_call_counter: unref user u call limit exceeded");
- else
- unref_peer(p, "update_call_counter: unref peer p, call limit exceeded");
+ ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", "peer", name, *call_limit);
+ unref_peer(p, "update_call_counter: unref peer p, call limit exceeded");
return -1;
}
}
@@ -4870,7 +4723,7 @@
ast_atomic_fetchadd_int(inuse, +1);
ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
if (sipdebug) {
- ast_debug(2, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
+ ast_debug(2, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", "peer", name, *inuse, *call_limit);
}
break;
@@ -4887,8 +4740,7 @@
if (p) {
ast_device_state_changed("SIP/%s", p->name);
unref_peer(p, "update_call_counter: unref_peer from call counter");
- } else /* u must be set */
- unref_user(u, "update_call_counter: unref_user from call counter");
+ }
return 0;
}
@@ -11119,7 +10971,10 @@
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);
- /* Saving TCP connections is useless, we won't be able to reconnect */
+ /* Saving TCP connections is useless, we won't be able to reconnect
+ XXX WHY???? XXX
+ \todo check this
+ */
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\nAddress: %s\r\nPort: %d\r\n", peer->name, ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port));
@@ -11451,7 +11306,7 @@
/*! \brief Change onhold state of a peer using a pvt structure */
static void sip_peer_hold(struct sip_pvt *p, int hold)
{
- struct sip_peer *peer = find_peer(p->peername, NULL, 1);
+ struct sip_peer *peer = find_peer(p->peername, NULL, 1, TRUE);
if (!peer)
return;
@@ -11518,7 +11373,7 @@
}
/*! \brief Send a fake 401 Unauthorized response when the administrator
- wants to hide the names of local users/peers from fishers
+ wants to hide the names of local devices from fishers
*/
static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, int reliable)
{
@@ -11600,7 +11455,7 @@
ast_string_field_set(p, exten, name);
build_contact(p);
- peer = find_peer(name, NULL, 1);
+ peer = find_peer(name, NULL, TRUE, TRUE);
if (!(peer && ast_apply_ha(peer->ha, sin))) {
/* Peer fails ACL check */
if (peer) {
@@ -11725,7 +11580,7 @@
case AUTH_USERNAME_MISMATCH:
/* Username and digest username does not match.
Asterisk uses the From: username for authentication. We need the
- users to use the same authentication user name until we support
+ devices to use the same authentication user name until we support
proper authentication by digest auth name */
transmit_response(p, "403 Authentication user name does not match account name", &p->initreq);
if (global_authfailureevents)
@@ -12476,118 +12331,6 @@
}
}
-/*! \brief Validate user authentication */
-static enum check_auth_result check_user_ok(struct sip_pvt *p, char *of,
- struct sip_request *req, int sipmethod, struct sockaddr_in *sin,
- enum xmittype reliable,
- char *rpid_num, char *calleridname, char *uri2)
-{
- enum check_auth_result res;
- struct sip_user *user = find_user(of, 1);
- int debug=sip_debug_test_addr(sin);
-
- /* Find user based on user name in the from header */
- if (!user) {
- if (debug)
- ast_verbose("No user '%s' in SIP users list\n", of);
- return AUTH_DONT_KNOW;
- }
- if (!ast_apply_ha(user->ha, sin)) {
- if (debug)
- ast_verbose("Found user '%s' for '%s', but fails host access\n",
- user->name, of);
- unref_user(user, "unref_user: from check_user_ok from find_user call, early return of AUTH_DONT_KNOW.");
- return AUTH_DONT_KNOW;
- }
- if (debug)
- ast_verbose("Found user '%s' for '%s'\n", user->name, of);
-
- 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);
- /* copy channel vars */
- p->chanvars = copy_vars(user->chanvars);
- p->prefs = user->prefs;
- /* Set Frame packetization */
- if (p->rtp) {
- ast_rtp_codec_setpref(p->rtp, &p->prefs);
- p->autoframing = user->autoframing;
- }
-
- 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))) {
- 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);
- /* Copy SIP extensions profile from INVITE */
- if (p->sipoptions)
- user->sipoptions = p->sipoptions;
-
- /* If we have a call limit, set flag */
- if (user->call_limit)
- ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
- if (!ast_strlen_zero(user->context))
- ast_string_field_set(p, context, user->context);
- if (!ast_strlen_zero(user->cid_num)) {
- char *tmp = ast_strdupa(user->cid_num);
- if (ast_is_shrinkable_phonenumber(tmp))
- ast_shrink_phone_number(tmp);
- ast_string_field_set(p, cid_num, tmp);
- }
- if (!ast_strlen_zero(user->cid_name))
- ast_string_field_set(p, cid_name, user->cid_name);
- ast_string_field_set(p, username, user->name);
- ast_string_field_set(p, peername, user->name);
- ast_string_field_set(p, peersecret, user->secret);
- ast_string_field_set(p, peermd5secret, user->md5secret);
- ast_string_field_set(p, subscribecontext, user->subscribecontext);
- ast_string_field_set(p, accountcode, user->accountcode);
- ast_string_field_set(p, language, user->language);
- ast_string_field_set(p, mohsuggest, user->mohsuggest);
- ast_string_field_set(p, mohinterpret, user->mohinterpret);
- ast_string_field_set(p, parkinglot, user->parkinglot);
- p->allowtransfer = user->allowtransfer;
- p->amaflags = user->amaflags;
- p->callgroup = user->callgroup;
- p->pickupgroup = user->pickupgroup;
- if (user->callingpres) /* User callingpres setting will override RPID header */
- p->callingpres = user->callingpres;
-
- /* Set default codec settings for this call */
- p->capability = user->capability; /* User codec choice */
- p->jointcapability = user->capability; /* Our codecs */
[... 1931 lines stripped ...]
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