[svn-commits] oej: branch oej/codename-pineapple r46380 - in
/team/oej/codename-pineapple/c...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Oct 28 10:57:10 MST 2006
Author: oej
Date: Sat Oct 28 12:57:09 2006
New Revision: 46380
URL: http://svn.digium.com/view/asterisk?rev=46380&view=rev
Log:
See, friends, see:
17222 74409 622008 channels/chan_sip.c
9739 40664 336765 channels/chan_sip3.c
Cutting, cutting, cutting....
Modified:
team/oej/codename-pineapple/channels/chan_sip3.c
team/oej/codename-pineapple/channels/sip3/sip3.h
team/oej/codename-pineapple/channels/sip3/sip3_dialog.c
team/oej/codename-pineapple/channels/sip3/sip3_parse.c
team/oej/codename-pineapple/channels/sip3/sip3_services.c
team/oej/codename-pineapple/channels/sip3/sip3funcs.h
Modified: team/oej/codename-pineapple/channels/chan_sip3.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/chan_sip3.c?rev=46380&r1=46379&r2=46380&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Sat Oct 28 12:57:09 2006
@@ -352,9 +352,6 @@
/*! \brief The peer list: Peers and Friends */
struct sip_device_list devicelist;
-/*! \brief The register list: Other SIP proxys we register with and place calls to */
-struct sip_register_list regl;
-
/* Forward declaration */
static int temp_pvt_init(void *data);
static void temp_pvt_cleanup(void *data);
@@ -409,17 +406,12 @@
static int does_peer_need_mwi(struct sip_peer *peer);
/*--- Dialog management */
-static struct sip_dialog *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
- int useglobal_nat, const int intended_method);
GNURK void sip_destroy(struct sip_dialog *p);
static void __sip_destroy(struct sip_dialog *p, int lockowner);
-static void __sip_ack(struct sip_dialog *p, int seqno, int resp, int sipmethod, int reset);
-static void __sip_pretend_ack(struct sip_dialog *p);
+static void __sip_ack(struct sip_dialog *dialog, int seqno, int resp, int sipmethod, int reset);
static int __sip_semi_ack(struct sip_dialog *p, int seqno, int resp, int sipmethod);
static int auto_congest(void *nothing);
static int update_call_counter(struct sip_dialog *fup, int event);
-static int hangup_sip2cause(int cause);
-static const char *hangup_cause2sip(int cause);
static void free_old_route(struct sip_route *route);
static void list_route(struct sip_route *route);
static void build_route(struct sip_dialog *p, struct sip_request *req, int backwards);
@@ -451,7 +443,6 @@
static int sip_refer_allocate(struct sip_dialog *p);
static void ast_quiet_chan(struct ast_channel *chan);
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
-static void do_setnat(struct sip_dialog *p, int natflags);
/*--- Device monitoring and Device/extension state handling */
static int cb_extensionstate(char *context, char* exten, int state, void *data);
@@ -502,18 +493,10 @@
static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey);
static void update_peer(struct sip_peer *p, int expiry);
-/*--- Internal UA client handling (outbound registrations) */
-static void sip_registry_destroy(struct sip_registry *reg);
-static int sip_reregister(void *data);
-static int __sip_do_register(struct sip_registry *r);
-static int sip_reg_timeout(void *data);
-static void sip_send_all_registers(void);
-
/*--- Parsing SIP requests and responses */
static void append_date(struct sip_request *req); /* Append date to SIP packet */
static int determine_firstline_parts(struct sip_request *req);
static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
-static char *get_in_brackets(char *tmp);
static void extract_uri(struct sip_dialog *p, struct sip_request *req);
static int get_refer_info(struct sip_dialog *transferer, struct sip_request *outgoing_req);
static int get_also_info(struct sip_dialog *p, struct sip_request *oreq);
@@ -527,26 +510,17 @@
static void free_old_route(struct sip_route *route);
/*--- Constructing requests and responses */
-static void initialize_initreq(struct sip_dialog *p, struct sip_request *req);
-static int init_req(struct sip_request *req, int sipmethod, const char *recip);
static int reqprep(struct sip_request *req, struct sip_dialog *p, int sipmethod, int seqno, int newbranch);
static void initreqprep(struct sip_request *req, struct sip_dialog *p, int sipmethod);
static int init_resp(struct sip_request *resp, const char *msg);
static int respprep(struct sip_request *resp, struct sip_dialog *p, const char *msg, const struct sip_request *req);
-static void build_via(struct sip_dialog *p);
static int create_addr_from_peer(struct sip_dialog *r, struct sip_peer *peer);
-static int create_addr(struct sip_dialog *dialog, const char *opeer);
-static char *generate_random_string(char *buf, size_t size);
-static void build_callid_pvt(struct sip_dialog *pvt);
-static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
-static void make_our_tag(char *tagbuf, size_t len);
static int add_text(struct sip_request *req, const char *text);
static int add_digit(struct sip_request *req, char digit);
static int add_vidupdate(struct sip_request *req);
static void add_route(struct sip_request *req, struct sip_route *route);
static void set_destination(struct sip_dialog *p, char *uri);
static void append_date(struct sip_request *req);
-static void build_contact(struct sip_dialog *p);
static void build_rpid(struct sip_dialog *p);
/*------Request handling functions */
@@ -568,7 +542,6 @@
static void handle_response_invite(struct sip_dialog *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response_refer(struct sip_dialog *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response_peerpoke(struct sip_dialog *p, int resp, struct sip_request *req);
-static int handle_response_register(struct sip_dialog *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response(struct sip_dialog *p, int resp, char *rest, struct sip_request *req, int seqno);
/*----- RTP interface functions */
@@ -605,7 +578,7 @@
/*! \brief Initialize the initital request packet in the pvt structure.
This packet is used for creating replies and future requests in
a dialog */
-static void initialize_initreq(struct sip_dialog *p, struct sip_request *req)
+GNURK void initialize_initreq(struct sip_dialog *p, struct sip_request *req)
{
if (p->initreq.headers && option_debug) {
ast_log(LOG_DEBUG, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
@@ -626,7 +599,7 @@
}
/*! \brief Build a Via header for a request */
-static void build_via(struct sip_dialog *p)
+GNURK void build_via(struct sip_dialog *p)
{
/* Work around buggy UNIDEN UIP200 firmware */
const char *rport = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : "";
@@ -660,21 +633,21 @@
}
/*! \brief Append to SIP dialog history with arg list */
-GNURK void append_history_full(struct sip_dialog *p, const char *fmt, ...)
+GNURK void append_history_full(struct sip_dialog *dialog, const char *fmt, ...)
{
va_list ap;
- if (!p)
+ if (!dialog)
return;
va_start(ap, fmt);
- append_history_va(p, fmt, ap);
+ append_history_va(dialog, fmt, ap);
va_end(ap);
return;
}
/*! \brief Acknowledges receipt of a packet and stops retransmission */
-static void __sip_ack(struct sip_dialog *p, int seqno, int resp, int sipmethod, int reset)
+static void __sip_ack(struct sip_dialog *dialog, int seqno, int resp, int sipmethod, int reset)
{
struct sip_pkt *cur, *prev = NULL;
@@ -684,19 +657,19 @@
msg = sip_method2txt(sipmethod);
- ast_mutex_lock(&p->lock);
- for (cur = p->packets; cur; prev = cur, cur = cur->next) {
+ ast_mutex_lock(&dialog->lock);
+ for (cur = dialog->packets; cur; prev = cur, cur = cur->next) {
if ((cur->seqno == seqno) && ((ast_test_flag(cur, FLAG_RESPONSE)) == resp) &&
((ast_test_flag(cur, FLAG_RESPONSE)) ||
(!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
- if (!resp && (seqno == p->pendinginvite)) {
+ if (!resp && (seqno == dialog->pendinginvite)) {
if (option_debug)
- ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
- p->pendinginvite = 0;
+ ast_log(LOG_DEBUG, "Acked pending invite %d\n", dialog->pendinginvite);
+ dialog->pendinginvite = 0;
}
/* this is our baby */
res = TRUE;
- UNLINK(cur, p->packets, prev);
+ UNLINK(cur, dialog->packets, prev);
if (cur->retransid > -1) {
if (sipdebug && option_debug > 3)
ast_log(LOG_DEBUG, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
@@ -707,26 +680,26 @@
break;
}
}
- ast_mutex_unlock(&p->lock);
+ ast_mutex_unlock(&dialog->lock);
if (option_debug)
- ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
-}
-
-/*! \brief Pretend to ack all packets
+ ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n", dialog->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
+}
+
+/*! \brief Pretend to ack all packets - nothing to do with SIP_ACK (the method)
* maybe the lock on p is not strictly necessary but there might be a race */
-static void __sip_pretend_ack(struct sip_dialog *p)
+GNURK void __sip_pretend_ack(struct sip_dialog *dialog)
{
struct sip_pkt *cur = NULL;
- while (p->packets) {
+ while (dialog->packets) {
int method;
- if (cur == p->packets) {
+ if (cur == dialog->packets) {
ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_method2txt(cur->method));
return;
}
- cur = p->packets;
+ cur = dialog->packets;
method = (cur->method) ? cur->method : find_sip_method(cur->data);
- __sip_ack(p, cur->seqno, ast_test_flag(cur, FLAG_RESPONSE), method, FALSE);
+ __sip_ack(dialog, cur->seqno, ast_test_flag(cur, FLAG_RESPONSE), method, FALSE);
}
}
@@ -756,7 +729,7 @@
}
-/*! \brief Copy SIP request, parse it */
+/*! \brief Copy SIP request, pre-parse it */
GNURK void parse_copy(struct sip_request *dst, const struct sip_request *src)
{
memset(dst, 0, sizeof(*dst));
@@ -773,67 +746,6 @@
snprintf(req->data + req->len, sizeof(req->data) - req->len, "\r\n");
req->len += strlen(req->data + req->len);
}
-}
-
-/*! \brief Locate closing quote in a string, skipping escaped quotes.
- * optionally with a limit on the search.
- * start must be past the first quote.
- */
-static const char *find_closing_quote(const char *start, const char *lim)
-{
- char last_char = '\0';
- const char *s;
- for (s = start; *s && s != lim; last_char = *s++) {
- if (*s == '"' && last_char != '\\')
- break;
- }
- return s;
-}
-
-/*! \brief Pick out text in brackets from character string
- \return pointer to terminated stripped string
- \param tmp input string that will be modified
- Examples:
-
- "foo" <bar> valid input, returns bar
- foo returns the whole string
- < "foo ... > returns the string between brackets
- < "foo... bogus (missing closing bracket), returns the whole string
- XXX maybe should still skip the opening bracket
- */
-static char *get_in_brackets(char *tmp)
-{
- const char *parse = tmp;
- char *first_bracket;
-
- /*
- * Skip any quoted text until we find the part in brackets.
- * On any error give up and return the full string.
- */
- while ( (first_bracket = strchr(parse, '<')) ) {
- char *first_quote = strchr(parse, '"');
-
- if (!first_quote || first_quote > first_bracket)
- break; /* no need to look at quoted part */
- /* the bracket is within quotes, so ignore it */
- parse = find_closing_quote(first_quote + 1, NULL);
- if (!*parse) { /* not found, return full string ? */
- /* XXX or be robust and return in-bracket part ? */
- ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp);
- break;
- }
- parse++;
- }
- if (first_bracket) {
- char *second_bracket = strchr(first_bracket + 1, '>');
- if (second_bracket) {
- *second_bracket = '\0';
- tmp = first_bracket + 1;
- } else {
- ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp);
- }
- }
- return tmp;
}
/*! \brief Send SIP MESSAGE text within a call
@@ -970,62 +882,62 @@
}
/*! \brief Update peer data in database (if used) */
-static void update_peer(struct sip_peer *p, int expiry)
-{
- int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
+static void update_peer(struct sip_peer *peer, int expiry)
+{
+ int rtcachefriends = ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
if (ast_test_flag(&global.flags[1], SIP_PAGE2_RTUPDATE) &&
- (ast_test_flag(&p->flags[0], SIP_REALTIME) || rtcachefriends)) {
- realtime_update_peer(p->name, &p->addr, p->username, rtcachefriends ? p->fullcontact : NULL, expiry);
- }
-}
-
-/*! \brief Support routine for find_peer */
+ (ast_test_flag(&peer->flags[0], SIP_REALTIME) || rtcachefriends)) {
+ realtime_update_peer(peer->name, &peer->addr, peer->username, rtcachefriends ? peer->fullcontact : NULL, expiry);
+ }
+}
+
+/*! \brief Support routine for find_device */
static int sip_addrcmp(char *name, struct sockaddr_in *sin)
{
/* We know name is the first field, so we can cast */
- struct sip_peer *p = (struct sip_peer *) name;
- return !(!inaddrcmp(&p->addr, sin) ||
- (ast_test_flag(&p->flags[0], SIP_INSECURE_PORT) &&
- (p->addr.sin_addr.s_addr == sin->sin_addr.s_addr)));
+ struct sip_peer *peer = (struct sip_peer *) name;
+ return !(!inaddrcmp(&peer->addr, sin) ||
+ (ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT) &&
+ (peer->addr.sin_addr.s_addr == sin->sin_addr.s_addr)));
}
/*! \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 */
-GNURK struct sip_peer *find_device(const char *peer, struct sockaddr_in *sin, int realtime)
-{
- struct sip_peer *p = NULL;
-
- if (peer)
- p = ASTOBJ_CONTAINER_FIND(&devicelist, peer);
+GNURK struct sip_peer *find_device(const char *device, struct sockaddr_in *sin, int realtime)
+{
+ struct sip_peer *peer = NULL;
+
+ if (device)
+ peer = ASTOBJ_CONTAINER_FIND(&devicelist, device);
else
- p = ASTOBJ_CONTAINER_FIND_FULL(&devicelist, sin, name, sip_addr_hashfunc, 1, sip_addrcmp);
-
- if (!p && realtime)
- p = realtime_peer(peer, sin);
-
- return p;
-}
-
-/*! \brief Set nat mode on the various data sockets */
-static void do_setnat(struct sip_dialog *p, int natflags)
+ peer = ASTOBJ_CONTAINER_FIND_FULL(&devicelist, sin, name, sip_addr_hashfunc, 1, sip_addrcmp);
+
+ if (!peer && realtime)
+ peer = realtime_peer(device, sin);
+
+ return peer;
+}
+
+/*! \brief Set nat mode on the various media streams */
+GNURK void do_setnat(struct sip_dialog *dialog, int natflags)
{
const char *mode = natflags ? "On" : "Off";
- if (p->rtp) {
+ if (dialog->rtp) {
if (option_debug)
ast_log(LOG_DEBUG, "Setting NAT on RTP to %s\n", mode);
- ast_rtp_setnat(p->rtp, natflags);
- }
- if (p->vrtp) {
+ ast_rtp_setnat(dialog->rtp, natflags);
+ }
+ if (dialog->vrtp) {
if (option_debug)
ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", mode);
- ast_rtp_setnat(p->vrtp, natflags);
- }
- if (p->udptl) {
+ ast_rtp_setnat(dialog->vrtp, natflags);
+ }
+ if (dialog->udptl) {
if (option_debug)
ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", mode);
- ast_udptl_setnat(p->udptl, natflags);
+ ast_udptl_setnat(dialog->udptl, natflags);
}
}
@@ -1133,7 +1045,7 @@
/*! \brief create address structure from peer 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_dialog *dialog, const char *opeer)
+GNURK int create_addr(struct sip_dialog *dialog, const char *opeer)
{
struct hostent *hp;
struct ast_hostent ahp;
@@ -1199,20 +1111,20 @@
/*! \brief Scheduled congestion on a call */
static int auto_congest(void *nothing)
{
- struct sip_dialog *p = nothing;
-
- ast_mutex_lock(&p->lock);
- p->initid = -1;
- if (p->owner) {
+ struct sip_dialog *dialog = nothing;
+
+ ast_mutex_lock(&dialog->lock);
+ dialog->initid = -1;
+ if (dialog->owner) {
/* XXX fails on possible deadlock */
- if (!ast_channel_trylock(p->owner)) {
- ast_log(LOG_NOTICE, "Auto-congesting %s\n", p->owner->name);
- append_history(p, "Cong", "Auto-congesting (timer)");
- ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- ast_channel_unlock(p->owner);
- }
- }
- ast_mutex_unlock(&p->lock);
+ if (!ast_channel_trylock(dialog->owner)) {
+ ast_log(LOG_NOTICE, "Auto-congesting %s\n", dialog->owner->name);
+ append_history(dialog, "Cong", "Auto-congesting (timer)");
+ ast_queue_control(dialog->owner, AST_CONTROL_CONGESTION);
+ ast_channel_unlock(dialog->owner);
+ }
+ }
+ ast_mutex_unlock(&dialog->lock);
return 0;
}
@@ -1293,32 +1205,6 @@
p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
}
return res;
-}
-
-/*! \brief Destroy registry object
- Objects created with the register= statement in static configuration */
-static void sip_registry_destroy(struct sip_registry *reg)
-{
- /* Really delete */
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname);
-
- if (reg->call) {
- /* Clear registry before destroying to ensure
- we don't get reentered trying to grab the registry lock */
- reg->call->registry = NULL;
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
- sip_destroy(reg->call);
- }
- if (reg->expire > -1)
- ast_sched_del(sched, reg->expire);
- if (reg->timeout > -1)
- ast_sched_del(sched, reg->timeout);
- ast_string_field_free_pools(reg);
- sipcounters.registry_objects--;
- free(reg);
-
}
/*! \brief Execute destruction of SIP dialog structure, release memory */
@@ -1520,167 +1406,6 @@
dialoglist_unlock();
}
-/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
-static int hangup_sip2cause(int cause)
-{
- /* Possible values taken from causes.h */
-
- switch(cause) {
- case 401: /* Unauthorized */
- return AST_CAUSE_CALL_REJECTED;
- case 403: /* Not found */
- return AST_CAUSE_CALL_REJECTED;
- case 404: /* Not found */
- return AST_CAUSE_UNALLOCATED;
- case 405: /* Method not allowed */
- return AST_CAUSE_INTERWORKING;
- case 407: /* Proxy authentication required */
- return AST_CAUSE_CALL_REJECTED;
- case 408: /* No reaction */
- return AST_CAUSE_NO_USER_RESPONSE;
- case 409: /* Conflict */
- return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
- case 410: /* Gone */
- return AST_CAUSE_UNALLOCATED;
- case 411: /* Length required */
- return AST_CAUSE_INTERWORKING;
- case 413: /* Request entity too large */
- return AST_CAUSE_INTERWORKING;
- case 414: /* Request URI too large */
- return AST_CAUSE_INTERWORKING;
- case 415: /* Unsupported media type */
- return AST_CAUSE_INTERWORKING;
- case 420: /* Bad extension */
- return AST_CAUSE_NO_ROUTE_DESTINATION;
- case 480: /* No answer */
- return AST_CAUSE_NO_ANSWER;
- case 481: /* No answer */
- return AST_CAUSE_INTERWORKING;
- case 482: /* Loop detected */
- return AST_CAUSE_INTERWORKING;
- case 483: /* Too many hops */
- return AST_CAUSE_NO_ANSWER;
- case 484: /* Address incomplete */
- return AST_CAUSE_INVALID_NUMBER_FORMAT;
- case 485: /* Ambigous */
- return AST_CAUSE_UNALLOCATED;
- case 486: /* Busy everywhere */
- return AST_CAUSE_BUSY;
- case 487: /* Request terminated */
- return AST_CAUSE_INTERWORKING;
- case 488: /* No codecs approved */
- return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
- case 491: /* Request pending */
- return AST_CAUSE_INTERWORKING;
- case 493: /* Undecipherable */
- return AST_CAUSE_INTERWORKING;
- case 500: /* Server internal failure */
- return AST_CAUSE_FAILURE;
- case 501: /* Call rejected */
- return AST_CAUSE_FACILITY_REJECTED;
- case 502:
- return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
- case 503: /* Service unavailable */
- return AST_CAUSE_CONGESTION;
- case 504: /* Gateway timeout */
- return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
- case 505: /* SIP version not supported */
- return AST_CAUSE_INTERWORKING;
- case 600: /* Busy everywhere */
- return AST_CAUSE_USER_BUSY;
- case 603: /* Decline */
- return AST_CAUSE_CALL_REJECTED;
- case 604: /* Does not exist anywhere */
- return AST_CAUSE_UNALLOCATED;
- case 606: /* Not acceptable */
- return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
- default:
- return AST_CAUSE_NORMAL;
- }
- /* Never reached */
- return 0;
-}
-
-/*! \brief Convert Asterisk hangup causes to SIP codes
-\verbatim
- Possible values from causes.h
- AST_CAUSE_NOTDEFINED AST_CAUSE_NORMAL AST_CAUSE_BUSY
- AST_CAUSE_FAILURE AST_CAUSE_CONGESTION AST_CAUSE_UNALLOCATED
-
- In addition to these, a lot of PRI codes is defined in causes.h
- ...should we take care of them too ?
-
- Quote RFC 3398
-
- ISUP Cause value SIP response
- ---------------- ------------
- 1 unallocated number 404 Not Found
- 2 no route to network 404 Not found
- 3 no route to destination 404 Not found
- 16 normal call clearing --- (*)
- 17 user busy 486 Busy here
- 18 no user responding 408 Request Timeout
- 19 no answer from the user 480 Temporarily unavailable
- 20 subscriber absent 480 Temporarily unavailable
- 21 call rejected 403 Forbidden (+)
- 22 number changed (w/o diagnostic) 410 Gone
- 22 number changed (w/ diagnostic) 301 Moved Permanently
- 23 redirection to new destination 410 Gone
- 26 non-selected user clearing 404 Not Found (=)
- 27 destination out of order 502 Bad Gateway
- 28 address incomplete 484 Address incomplete
- 29 facility rejected 501 Not implemented
- 31 normal unspecified 480 Temporarily unavailable
-\endverbatim
-*/
-static const char *hangup_cause2sip(int cause)
-{
- switch (cause) {
- case AST_CAUSE_UNALLOCATED: /* 1 */
- case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
- case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
- return "404 Not Found";
- case AST_CAUSE_CONGESTION: /* 34 */
- case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
- return "503 Service Unavailable";
- case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
- return "408 Request Timeout";
- case AST_CAUSE_NO_ANSWER: /* 19 */
- return "480 Temporarily unavailable";
- case AST_CAUSE_CALL_REJECTED: /* 21 */
- return "403 Forbidden";
- case AST_CAUSE_NUMBER_CHANGED: /* 22 */
- return "410 Gone";
- case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
- return "480 Temporarily unavailable";
- case AST_CAUSE_INVALID_NUMBER_FORMAT:
- return "484 Address incomplete";
- case AST_CAUSE_USER_BUSY:
- return "486 Busy here";
- case AST_CAUSE_FAILURE:
- return "500 Server internal failure";
- case AST_CAUSE_FACILITY_REJECTED: /* 29 */
- return "501 Not Implemented";
- case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
- return "503 Service Unavailable";
- /* Used in chan_iax2 */
- case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
- return "502 Bad Gateway";
- case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
- return "488 Not Acceptable Here";
-
- case AST_CAUSE_NOTDEFINED:
- default:
- if (option_debug)
- ast_log(LOG_DEBUG, "AST hangup cause %d (no match found in SIP)\n", cause);
- return NULL;
- }
-
- /* Never reached */
- return 0;
-}
-
-
/*! \brief sip_hangup: Hangup SIP call
* Part of PBX interface, called from ast_hangup */
static int sip_hangup(struct ast_channel *ast)
@@ -2303,7 +2028,7 @@
}
/*! \brief Generate 32 byte random string for callid's etc */
-static char *generate_random_string(char *buf, size_t size)
+GNURK char *generate_random_string(char *buf, size_t size)
{
long val[4];
int x;
@@ -2316,7 +2041,7 @@
}
/*! \brief Build SIP Call-ID value for a non-REGISTER transaction */
-static void build_callid_pvt(struct sip_dialog *pvt)
+GNURK void build_callid_pvt(struct sip_dialog *pvt)
{
char buf[33];
@@ -2324,147 +2049,6 @@
ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
-}
-
-/*! \brief Build SIP Call-ID value for a REGISTER transaction */
-static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain)
-{
- char buf[33];
-
- const char *host = S_OR(fromdomain, ast_inet_ntoa(ourip));
-
- ast_string_field_build(reg, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
-}
-
-/*! \brief Make our SIP dialog tag */
-static void make_our_tag(char *tagbuf, size_t len)
-{
- if (sipdebug)
- snprintf(tagbuf, len, "asterisk%08lx", ast_random());
- else
- snprintf(tagbuf, len, "%08lx", ast_random());
-}
-
-/*! \brief Allocate SIP_PVT structure and set defaults */
-static struct sip_dialog *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
- int useglobal_nat, const int intended_method)
-{
- struct sip_dialog *p;
-
- if (!(p = ast_calloc(1, sizeof(*p))))
- return NULL;
-
- if (ast_string_field_init(p, 512)) {
- free(p);
- return NULL;
- }
-
- ast_mutex_init(&p->lock);
-
- p->method = intended_method;
- p->initid = -1;
- p->autokillid = -1;
- p->subscribed = NONE;
- p->stateid = -1;
- p->prefs = global.default_prefs; /* Set default codecs for this call */
-
- if (intended_method != SIP_OPTIONS) /* Peerpoke has it's own system */
- p->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
-
- if (sin) {
- p->sa = *sin;
- if (sip_ouraddrfor(&p->sa.sin_addr, &p->ourip))
- p->ourip = sipnet.__ourip;
- } else
- p->ourip = sipnet.__ourip;
-
- /* Copy global flags to this PVT at setup. */
- ast_copy_flags(&p->flags[0], &global.flags[0], SIP_FLAGS_TO_COPY);
- ast_copy_flags(&p->flags[1], &global.flags[1], SIP_PAGE2_FLAGS_TO_COPY);
-
- ast_set2_flag(&p->flags[0], !global.recordhistory, SIP_NO_HISTORY);
-
- p->branch = ast_random();
- make_our_tag(p->tag, sizeof(p->tag));
- p->ocseq = INITIAL_CSEQ;
-
- if (sip_method_needrtp(intended_method)) {
- p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, sipnet.bindaddr.sin_addr);
- /* If the global videosupport flag is on, we always create a RTP interface for video */
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT))
- p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, sipnet.bindaddr.sin_addr);
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT))
- p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, sipnet.bindaddr.sin_addr);
- if (!p->rtp || (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)) {
- ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n",
- ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video" : "", strerror(errno));
- ast_mutex_destroy(&p->lock);
- if (p->chanvars) {
- ast_variables_destroy(p->chanvars);
- p->chanvars = NULL;
- }
- free(p);
- return NULL;
- }
- ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
- ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- ast_rtp_settos(p->rtp, global.tos_audio);
- if (p->vrtp) {
- ast_rtp_settos(p->vrtp, global.tos_video);
- ast_rtp_setdtmf(p->vrtp, 0);
- ast_rtp_setdtmfcompensate(p->vrtp, 0);
- }
- if (p->udptl)
- ast_udptl_settos(p->udptl, global.tos_audio);
- p->rtptimeout = global.rtptimeout;
- p->rtpholdtimeout = global.rtpholdtimeout;
- p->rtpkeepalive = global.rtpkeepalive;
- p->maxcallbitrate = global.default_maxcallbitrate;
- }
-
- if (useglobal_nat && sin) {
- /* Setup NAT structure according to global settings if we have an address */
- ast_copy_flags(&p->flags[0], &global.flags[0], SIP_NAT);
- p->recv = *sin;
- do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
- }
-
- if (p->method != SIP_REGISTER)
- ast_string_field_set(p, fromdomain, global.default_fromdomain);
- build_via(p);
- if (!callid)
- build_callid_pvt(p);
- else
- ast_string_field_set(p, callid, callid);
- /* Assign default music on hold class */
- ast_string_field_set(p, mohinterpret, global.default_mohinterpret);
- ast_string_field_set(p, mohsuggest, global.default_mohsuggest);
- p->capability = global.capability;
- p->allowtransfer = global.allowtransfer;
- if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
- (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
- p->noncodeccapability |= AST_RTP_DTMF;
- if (p->udptl) {
- p->t38.capability = global.t38_capability;
- if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY)
- p->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
- else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_FEC)
- p->t38.capability |= T38FAX_UDP_EC_FEC;
- else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_NONE)
- p->t38.capability |= T38FAX_UDP_EC_NONE;
- p->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
- p->t38.jointcapability = p->t38.capability;
- }
- ast_string_field_set(p, context, global.default_context);
-
- /* Add to active dialog list */
- dialoglist_lock();
- p->next = dialoglist;
- dialoglist = p;
- dialoglist_unlock();
- if (option_debug)
- ast_log(LOG_DEBUG, "Allocating new SIP dialog for %s - %s (%s)\n", callid ? callid : "(No Call-ID)", sip_method2txt(intended_method), p->rtp ? "With RTP" : "No RTP");
- return p;
}
/*! \brief Connect incoming SIP message to current dialog or create new dialog structure
@@ -2555,128 +2139,6 @@
}
return p;
-}
-
-/*! \brief Parse register=> line in sip.conf and add to registry */
-GNURK int sip_register(char *value, int lineno)
-{
- struct sip_registry *reg;
- char copy[256];
- char *username=NULL, *hostname=NULL, *secret=NULL, *authuser=NULL;
- char *porta=NULL;
- char *contact=NULL;
- char *stringp=NULL;
-
- if (!value)
- return -1;
- ast_copy_string(copy, value, sizeof(copy));
- stringp=copy;
- username = stringp;
- hostname = strrchr(stringp, '@');
- if (hostname)
- *hostname++ = '\0';
- if (ast_strlen_zero(username) || ast_strlen_zero(hostname)) {
- ast_log(LOG_WARNING, "Format for registration is user[:secret[:authuser]]@host[:port][/contact] at line %d\n", lineno);
- return -1;
- }
- stringp = username;
- username = strsep(&stringp, ":");
- if (username) {
- secret = strsep(&stringp, ":");
- if (secret)
- authuser = strsep(&stringp, ":");
- }
- stringp = hostname;
- hostname = strsep(&stringp, "/");
- if (hostname)
- contact = strsep(&stringp, "/");
- if (ast_strlen_zero(contact))
- contact = "s";
- stringp=hostname;
- hostname = strsep(&stringp, ":");
- porta = strsep(&stringp, ":");
-
- if (porta && !atoi(porta)) {
- ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno);
- return -1;
- }
- if (!(reg = ast_calloc(1, sizeof(*reg)))) {
- ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n");
- return -1;
- }
-
- if (ast_string_field_init(reg, 256)) {
- ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry strings\n");
- free(reg);
- return -1;
- }
-
- sipcounters.registry_objects++;
- ASTOBJ_INIT(reg);
- ast_string_field_set(reg, contact, contact);
- if (username)
- ast_string_field_set(reg, username, username);
- if (hostname)
- ast_string_field_set(reg, hostname, hostname);
- if (authuser)
- ast_string_field_set(reg, authuser, authuser);
- if (secret)
- ast_string_field_set(reg, secret, secret);
- reg->expire = -1;
- reg->expiry = expiry.default_expiry;
- reg->timeout = -1;
- reg->refresh = expiry.default_expiry;
- reg->portno = porta ? atoi(porta) : 0;
- reg->callid_valid = FALSE;
- reg->ocseq = INITIAL_CSEQ;
- ASTOBJ_CONTAINER_LINK(®l, reg); /* Add the new registry entry to the list */
- ASTOBJ_UNREF(reg,sip_registry_destroy);
- return 0;
-}
-
-/*! \brief Parse multiline SIP headers into one header */
-GNURK int lws2sws(char *msgbuf, int len)
-{
- int h = 0, t = 0;
- int lws = 0;
-
- for (; h < len;) {
- /* Eliminate all CRs */
- if (msgbuf[h] == '\r') {
- h++;
- continue;
- }
- /* Check for end-of-line */
- if (msgbuf[h] == '\n') {
- /* Check for end-of-message */
- if (h + 1 == len)
- break;
- /* Check for a continuation line */
- if (msgbuf[h + 1] == ' ' || msgbuf[h + 1] == '\t') {
- /* Merge continuation line */
- h++;
- continue;
- }
- /* Propagate LF and start new line */
- msgbuf[t++] = msgbuf[h++];
- lws = 0;
- continue;
- }
- if (msgbuf[h] == ' ' || msgbuf[h] == '\t') {
- if (lws) {
- h++;
- continue;
- }
- msgbuf[t++] = msgbuf[h++];
- lws = 1;
- continue;
- }
- msgbuf[t++] = msgbuf[h++];
- if (lws)
- lws = 0;
- }
- msgbuf[t] = '\0';
- return t;
}
/*! \brief Parse a SIP message
@@ -2899,7 +2361,7 @@
}
/*! \brief Initialize SIP request */
-static int init_req(struct sip_request *req, int sipmethod, const char *recip)
+GNURK int init_req(struct sip_request *req, int sipmethod, const char *recip)
{
/* Initialize a request */
memset(req, 0, sizeof(*req));
@@ -3568,7 +3030,7 @@
}
/*! \brief Build contact header - the contact header we send out */
-static void build_contact(struct sip_dialog *p)
+GNURK void build_contact(struct sip_dialog *p)
{
/* Construct Contact: header */
if (sipnet_ourport() != STANDARD_SIP_PORT) /* Needs to be 5060, according to the RFC */
@@ -4114,284 +3576,6 @@
return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}
-/*! \brief Convert registration state status to string */
-GNURK char *regstate2str(enum sipregistrystate regstate)
-{
- switch(regstate) {
- case REG_STATE_FAILED:
- return "Failed";
- case REG_STATE_UNREGISTERED:
- return "Unregistered";
- case REG_STATE_REGSENT:
- return "Request Sent";
- case REG_STATE_AUTHSENT:
- return "Auth. Sent";
- case REG_STATE_REGISTERED:
- return "Registered";
- case REG_STATE_REJECTED:
- return "Rejected";
- case REG_STATE_TIMEOUT:
- return "Timeout";
- case REG_STATE_NOAUTH:
- return "No Authentication";
- default:
- return "Unknown";
- }
-}
-
-/*! \brief Update registration with SIP Proxy */
-static int sip_reregister(void *data)
-{
- /* if we are here, we know that we need to reregister. */
- struct sip_registry *r= ASTOBJ_REF((struct sip_registry *) data);
-
- /* if we couldn't get a reference to the registry object, punt */
- if (!r)
- return 0;
-
- if (r->call && !ast_test_flag(&r->call->flags[0], SIP_NO_HISTORY))
- append_history(r->call, "RegistryRenew", "Account: %s@%s", r->username, r->hostname);
- /* Since registry's are only added/removed by the the monitor thread, this
- may be overkill to reference/dereference at all here */
- if (sipdebug)
- ast_log(LOG_NOTICE, " -- Re-registration for %s@%s\n", r->username, r->hostname);
-
- r->expire = -1;
- __sip_do_register(r);
- ASTOBJ_UNREF(r, sip_registry_destroy);
- return 0;
-}
-
-/*! \brief Register with SIP proxy */
-static int __sip_do_register(struct sip_registry *r)
-{
- int res;
-
- res = transmit_register(r, SIP_REGISTER, NULL, NULL);
- return res;
-}
-
-/*! \brief Registration timeout, register again */
-static int sip_reg_timeout(void *data)
-{
-
- /* if we are here, our registration timed out, so we'll just do it over */
- struct sip_registry *r = ASTOBJ_REF((struct sip_registry *) data);
- struct sip_dialog *p;
- int res;
-
- /* if we couldn't get a reference to the registry object, punt */
- if (!r)
- return 0;
-
- ast_log(LOG_NOTICE, " -- Registration for '%s@%s' timed out, trying again (Attempt #%d)\n", r->username, r->hostname, r->regattempts);
- if (r->call) {
- /* Unlink us, destroy old call. Locking is not relevant here because all this happens
- in the single SIP manager thread. */
- p = r->call;
- if (p->registry)
- ASTOBJ_UNREF(p->registry, sip_registry_destroy);
- r->call = NULL;
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- /* Pretend to ACK anything just in case */
- __sip_pretend_ack(p); /* XXX we need p locked, not sure we have */
- }
- /* If we have a limit, stop registration and give up */
- if (global.regattempts_max && (r->regattempts > global.regattempts_max)) {
- /* Ok, enough is enough. Don't try any more */
- /* We could add an external notification here...
- steal it from app_voicemail :-) */
- ast_log(LOG_NOTICE, " -- Giving up forever trying to register '%s@%s'\n", r->username, r->hostname);
- r->regstate = REG_STATE_FAILED;
- } else {
- r->regstate = REG_STATE_UNREGISTERED;
- r->timeout = -1;
- res=transmit_register(r, SIP_REGISTER, NULL, NULL);
- }
- manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelDriver: SIP\r\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate));
- ASTOBJ_UNREF(r, sip_registry_destroy);
- return 0;
-}
-
-/*! \brief Transmit register to SIP proxy or UA */
-GNURK int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader)
-{
- struct sip_request req;
- char from[256];
- char to[256];
- char tmp[80];
- char addr[80];
- struct sip_dialog *p;
-
- /* exit if we are already in process with this registrar ?*/
- if ( r == NULL || ((auth==NULL) && (r->regstate==REG_STATE_REGSENT || r->regstate==REG_STATE_AUTHSENT))) {
- ast_log(LOG_NOTICE, "Strange, trying to register %s@%s when registration already pending\n", r->username, r->hostname);
- return 0;
- }
-
- if (r->call) { /* We have a registration */
- if (!auth) {
- ast_log(LOG_WARNING, "Already have a REGISTER going on to %s@%s?? \n", r->username, r->hostname);
- return 0;
- } else {
- p = r->call;
- make_our_tag(p->tag, sizeof(p->tag)); /* create a new local tag for every register attempt */
- ast_string_field_free(p, theirtag); /* forget their old tag, so we don't match tags when getting response */
- }
- } else {
- /* Build callid for registration if we haven't registered before */
- if (!r->callid_valid) {
- build_callid_registry(r, sipnet.__ourip, global.default_fromdomain);
- r->callid_valid = TRUE;
- }
- /* Allocate SIP packet for registration */
- if (!(p = sip_alloc( r->callid, NULL, 0, SIP_REGISTER))) {
- ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n");
- return 0;
- }
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
- append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
- /* Find address to hostname */
- if (create_addr(p, r->hostname)) {
- /* we have what we hope is a temporary network error,
- * probably DNS. We need to reschedule a registration try */
- sip_destroy(p);
- if (r->timeout > -1) {
- ast_sched_del(sched, r->timeout);
- r->timeout = ast_sched_add(sched, global.reg_timeout*1000, sip_reg_timeout, r);
- ast_log(LOG_WARNING, "Still have a registration timeout for %s@%s (create_addr() error), %d\n", r->username, r->hostname, r->timeout);
- } else {
- r->timeout = ast_sched_add(sched, global.reg_timeout*1000, sip_reg_timeout, r);
- ast_log(LOG_WARNING, "Probably a DNS error for registration to %s@%s, trying REGISTER again (after %d seconds)\n", r->username, r->hostname, global.reg_timeout);
- }
- r->regattempts++;
- return 0;
- }
[... 1400 lines stripped ...]
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