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/*
* Asterisk -- A telephony toolkit for Linux.
*
* Implementation of Session Initiation Protocol
*
* Copyright (C) 2004, Digium, Inc.
* Mark Spencer <markster@digium.com>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*/
#include <string.h>
#include <asterisk/lock.h>
#include <asterisk/channel.h>
#include <asterisk/channel_pvt.h>
#include <asterisk/config.h>
#include <asterisk/logger.h>
#include <asterisk/module.h>
#include <asterisk/pbx.h>
#include <asterisk/options.h>
#include <asterisk/lock.h>
#include <asterisk/sched.h>
#include <asterisk/io.h>
#include <asterisk/rtp.h>
#include <asterisk/acl.h>
#include <asterisk/manager.h>
#include <asterisk/cli.h>
#include <asterisk/md5.h>
#include <asterisk/app.h>
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#include <asterisk/features.h>
#include <asterisk/srv.h>
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#include <asterisk/causes.h>
#include <asterisk/utils.h>
#ifdef OSP_SUPPORT
#include <asterisk/astosp.h>
#endif
#include <errno.h>
#include <unistd.h>
#include <stdlib.h>
#include <fcntl.h>
#include <netdb.h>
#include <arpa/inet.h>
#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX"
#endif
#define VIDEO_CODEC_MASK 0x1fc0000 /* Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
#ifndef IPTOS_MINCOST
#define IPTOS_MINCOST 0x02
#endif
#define DEFAULT_DEFAULT_EXPIRY 120
#define DEFAULT_MAX_EXPIRY 3600
/* guard limit must be larger than guard secs */
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/* guard min must be < 1000, and should be >= 250 */
#define EXPIRY_GUARD_SECS 15 /* How long before expiry do we reregister */
#define EXPIRY_GUARD_LIMIT 30 /* Below here, we use EXPIRY_GUARD_PCT instead of EXPIRY_GUARD_SECS */
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#define EXPIRY_GUARD_MIN 500 /* This is the minimum guard time applied. If GUARD_PCT turns out
to be lower than this, it will use this time instead. This is in
milliseconds. */
#define EXPIRY_GUARD_PCT 0.20 /* Percentage of expires timeout to use when below EXPIRY_GUARD_LIMIT */
#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
#endif
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#define CALLERID_UNKNOWN "Unknown"
/* --- Choices for DTMF support in SIP channel */
#define SIP_DTMF_RFC2833 (1 << 0)
#define SIP_DTMF_INBAND (1 << 1)
#define SIP_DTMF_INFO (1 << 2)
static int max_expiry = DEFAULT_MAX_EXPIRY;
static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
#define DEFAULT_MAXMS 2000 /* Must be faster than 2 seconds by default */
#define DEFAULT_FREQ_OK 60 * 1000 /* How often to check for the host to be up */
#define DEFAULT_FREQ_NOTOK 10 * 1000 /* How often to check, if the host is down... */
#define DEFAULT_RETRANS 1000 /* How frequently to retransmit */
#define MAX_RETRANS 5 /* Try only 5 times for retransmissions */
/* SIP Debug */
#define DEBUG_READ 0 /* Recieved data */
#define DEBUG_SEND 1 /* Transmit data */
static char *desc = "Session Initiation Protocol (SIP)";
static char *type = "SIP";
static char *tdesc = "Session Initiation Protocol (SIP)";
static char *config = "sip.conf";
#define DEFAULT_SIP_PORT 5060 /* From RFC 2543 */
#define SIP_MAX_PACKET 4096 /* Also from RFC 2543, should sub headers tho */
#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER"
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static char default_useragent[AST_MAX_EXTENSION] = DEFAULT_USERAGENT;
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static char default_context[AST_MAX_EXTENSION] = "default";
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static char default_language[MAX_LANGUAGE] = "";
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static char default_callerid[AST_MAX_EXTENSION] = "asterisk";
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static char default_fromdomain[AST_MAX_EXTENSION] = "";
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static char notifymime[AST_MAX_EXTENSION] = "application/simple-message-summary";
static int srvlookup = 0;
static int pedanticsipchecking = 0;
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static int global_rtptimeout = 0;
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static int global_rtpholdtimeout = 0;
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static int global_trustrpid = 0;
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static int global_progressinband = 0;
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static int global_ospauth = 0;
AST_MUTEX_DEFINE_STATIC(usecnt_lock);
/* Protect the interface list (of sip_pvt's) */
/* 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(netlock);
AST_MUTEX_DEFINE_STATIC(monlock);
/* This is the thread for the monitor which checks for input on the channels
which are not currently in use. */
static pthread_t monitor_thread = AST_PTHREADT_NULL;
/* Codecs that we support by default: */
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static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
static struct sockaddr_in debugaddr;
static int videosupport = 0;
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static int global_dtmfmode = SIP_DTMF_RFC2833; /* DTMF mode default */
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static int recordhistory = 0;
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static int global_promiscredir;
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static char global_musicclass[MAX_LANGUAGE] = ""; /* Global music on hold class */
static char global_realm[AST_MAX_EXTENSION] = "asterisk"; /* Default realm */
static char regcontext[AST_MAX_EXTENSION] = "";
static struct sched_context *sched;
static struct io_context *io;
/* The private structures of the sip channels are linked for
selecting outgoing channels */
#define SIP_MAX_HEADERS 64
#define SIP_MAX_LINES 64
#define DEC_IN_USE 0
#define INC_IN_USE 1
#define DEC_OUT_USE 2
#define INC_OUT_USE 3
static struct sip_codec_pref {
int codec;
struct sip_codec_pref *next;
} *prefs;
/* sip_request: The data grabbed from the UDP socket */
char *rlPart1; /* SIP Method Name or "SIP/2.0" protocol version */
char *rlPart2; /* The Request URI or Response Status */
int len;
int headers; /* SIP Headers */
char *header[SIP_MAX_HEADERS];
int lines; /* SDP Content */
char *line[SIP_MAX_LINES];
char data[SIP_MAX_PACKET];
};
struct sip_route {
struct sip_route *next;
char hop[0];
};
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struct sip_history {
char event[80];
struct sip_history *next;
};
/* sip_pvt: PVT structures are used for each SIP conversation, ie. a call */
ast_mutex_t lock; /* Channel private lock */
char callid[80]; /* Global CallID */
char randdata[80]; /* Random data */
unsigned int ocseq; /* Current outgoing seqno */
unsigned int icseq; /* Current incoming seqno */
unsigned int callgroup; /* Call group */
unsigned int pickupgroup; /* Pickup group */
int lastinvite; /* Last Cseq of invite */
int alreadygone; /* Whether or not we've already been destroyed by or peer */
int needdestroy; /* if we need to be destroyed */
int capability; /* Special capability (codec) */
int novideo; /* Didn't get video in invite, don't offer */
int jointcapability; /* Supported capability at both ends (codecs ) */
int peercapability; /* Supported peer capability */
int prefcodec; /* Preferred codec (outbound only) */
int callingpres; /* Calling presentation */
int outgoing; /* Outgoing or incoming call? */
int authtries; /* Times we've tried to authenticate */
int insecure; /* Don't check source port/ip */
int expiry; /* How long we take to expire */
int branch; /* One random number */
int canreinvite; /* Do we support reinvite */
int ringing; /* Have sent 180 ringing */
int progress; /* Have sent 183 message progress */
int useclientcode; /* Trust X-ClientCode info message */
int tag; /* Another random number */
int nat; /* Whether to try to support NAT */
int sessionid; /* SDP Session ID */
int sessionversion; /* SDP Session Version */
struct sockaddr_in sa; /* Our peer */
struct sockaddr_in redirip; /* Where our RTP should be going if not to us */
struct sockaddr_in vredirip; /* Where our Video RTP should be going if not to us */
int redircodecs; /* Redirect codecs */
struct sockaddr_in recv; /* Received as */
struct in_addr ourip; /* Our IP */
struct ast_channel *owner; /* Who owns us */
char exten[AST_MAX_EXTENSION]; /* Extension where to start */
char refer_to[AST_MAX_EXTENSION]; /* Place to store REFER-TO extension */
char referred_by[AST_MAX_EXTENSION]; /* Place to store REFERRED-BY extension */
char refer_contact[AST_MAX_EXTENSION]; /* Place to store Contact info from a REFER extension */
struct sip_pvt *refer_call; /* Call we are referring */
struct sip_route *route; /* Head of linked list of routing steps (fm Record-Route) */
int route_persistant; /* Is this the "real" route? */
char from[256]; /* The From: header */
char useragent[256]; /* User agent in SIP request */
char context[AST_MAX_EXTENSION]; /* Context for this call */
char fromdomain[AST_MAX_EXTENSION]; /* Domain to show in the from field */
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char fromuser[AST_MAX_EXTENSION]; /* Domain to show in the user field */
char tohost[AST_MAX_EXTENSION]; /* Host we should put in the "to" field */
char language[MAX_LANGUAGE]; /* Default language for this call */
char musicclass[MAX_LANGUAGE]; /* Music on Hold class */
char rdnis[256]; /* Referring DNIS */
char theirtag[256]; /* Their tag */
char username[256];
char peername[256];
char authname[256]; /* Who we use for authentication */
char uri[256]; /* Original requested URI */
char cid_num[256]; /* Caller*ID */
char cid_name[256]; /* Caller*ID */
char fullcontact[128]; /* Extra parameters to go in the "To" header */
char accountcode[20]; /* Account code */
char our_contact[256]; /* Our contact header */
char realm[256]; /* Authorization realm */
char nonce[256]; /* Authorization nonce */
char opaque[256]; /* Opaque nonsense */
char qop[80]; /* Quality of Protection, since SIP wasn't complicated enough yet. */
char domain[256]; /* Authorization nonce */
char lastmsg[256]; /* Last Message sent/received */
int amaflags; /* AMA Flags */
int pendinginvite; /* Any pending invite */
int needreinvite; /* Do we need to send another reinvite? */
int pendingbye; /* Need to send bye after we ack? */
int gotrefer; /* Got a refer? */
int ospauth; /* Allow OSP Authentication */
int osphandle; /* OSP Handle for call */
time_t ospstart; /* OSP Start time */
#endif
struct sip_request initreq; /* Initial request */
int maxtime; /* Max time for first response */
int initid; /* Auto-congest ID if appropriate */
int autokillid; /* Auto-kill ID */
time_t lastrtprx; /* Last RTP received */
int rtptimeout; /* RTP timeout time */
int rtpholdtimeout; /* RTP timeout when on hold */
int subscribed;
int stateid;
int dialogver;
int promiscredir; /* Promiscuous redirection */
int trustrpid;
int progressinband;
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int dtmfmode;
struct ast_dsp *vad;
struct sip_peer *peerpoke; /* If this calls is to poke a peer, which one */
struct sip_registry *registry; /* If this is a REGISTER call, to which registry */
struct ast_rtp *rtp; /* RTP Session */
struct ast_rtp *vrtp; /* Video RTP session */
struct sip_pkt *packets; /* Packets scheduled for re-transmission */
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struct sip_history *history; /* History of this SIP dialog */
struct sip_pvt *next; /* Next call in chain */
#define FLAG_RESPONSE (1 << 0)
#define FLAG_FATAL (1 << 1)
/* sip packet - read in sipsock_read, transmitted in send_request */
struct sip_pkt {
struct sip_pkt *next; /* Next packet */
int retrans; /* Retransmission number */
int seqno; /* Sequence number */
int flags; /* non-zero if this is a response packet (e.g. 200 OK) */
struct sip_pvt *owner; /* Owner call */
int retransid; /* Retransmission ID */
int packetlen; /* Length of packet */
char data[0];
};
/* 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];
char cid_num[80];
char cid_name[80];
char musicclass[MAX_LANGUAGE]; /* Music on Hold class */
char useragent[256]; /* User agent in SIP request */
unsigned int callgroup;
unsigned int pickupgroup;
#ifdef OSP_SUPPORT
int ospauth; /* Allow OSP Authentication */
#endif
int dtmfmode;
int inUse;
int incominglimit;
int useclientcode;
/* Structure for SIP peer data, we place calls to peers if registred or fixed IP address (host) */
struct sip_peer {
char name[80];
char secret[80];
char context[80]; /* JK02: peers need context too to allow parking etc */
char regexten[AST_MAX_EXTENSION]; /* Extension to register (if regcontext is used) */
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char fromuser[80];
char fromdomain[80];
char cid_num[80];
char cid_name[80];
char language[MAX_LANGUAGE];
char musicclass[MAX_LANGUAGE]; /* Music on Hold class */
char useragent[256]; /* User agent in SIP request */
int lastmsgssent;
time_t lastmsgcheck;
int rtptimeout;
int rtpholdtimeout;
#ifdef OSP_SUPPORT
int ospauth; /* Allow OSP Authentication */
#endif
unsigned int callgroup;
unsigned int pickupgroup;
int dtmfmode;
int trustrpid;
int useclientcode;
struct sockaddr_in addr;
struct in_addr mask;
/* Qualification */
struct sip_pvt *call; /* Call pointer */
int pokeexpire; /* When to expire poke */
int lastms; /* How long last response took (in ms), or -1 for no response */
int maxms; /* Max ms we will accept for the host to be up, 0 to not monitor */
struct timeval ps; /* Ping send time */
struct sockaddr_in defaddr;
struct ast_ha *ha;
int delme;
AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
static int sip_reloading = 0;
#define REG_STATE_UNREGISTERED 0
#define REG_STATE_REGSENT 1
#define REG_STATE_AUTHSENT 2
#define REG_STATE_REGISTERED 3
#define REG_STATE_REJECTED 4
#define REG_STATE_TIMEOUT 5
#define REG_STATE_NOAUTH 6
#define SIP_NAT_NEVER 0
#define SIP_NAT_RFC3581 (1 << 0)
#define SIP_NAT_ROUTE (1 << 2)
#define SIP_NAT_ALWAYS (SIP_NAT_ROUTE | SIP_NAT_RFC3581)
/* sip_registry: Registrations with other SIP proxies */
int portno; /* Optional port override */
char username[80]; /* Who we are registering as */
char authuser[80]; /* Who we *authenticate* as */
char secret[80]; /* Password or key name in []'s */
char contact[80]; /* Contact extension */
int expire; /* Sched ID of expiration */
int timeout; /* sched id of sip_reg_timeout */
int refresh; /* How often to refresh */
struct sip_pvt *call; /* create a sip_pvt structure for each outbound "registration call" in progress */
int regstate;
int callid_valid; /* 0 means we haven't chosen callid for this registry yet. */
char callid[80]; /* Global CallID for this registry */
unsigned int ocseq; /* Sequence number we got to for REGISTERs for this registry */
struct sockaddr_in us; /* Who the server thinks we are */
/*--- The user list: Users and friends ---*/
static struct ast_user_list {
struct sip_user *users;
ast_mutex_t lock;
/*--- The peer list: Peers and Friends ---*/
static struct ast_peer_list {
struct sip_peer *peers;
ast_mutex_t lock;
/*--- The register list: Other SIP proxys we register with and call ---*/
static struct ast_register_list {
struct sip_registry *registrations;
ast_mutex_t lock;
#define REINVITE_INVITE 1
#define REINVITE_UPDATE 2
static int __sip_do_register(struct sip_registry *r);
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static int global_nat = SIP_NAT_RFC3581;
static int global_canreinvite = REINVITE_INVITE;
static struct sockaddr_in externip;
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static struct ast_ha *localaddr;
static struct ast_frame *sip_read(struct ast_channel *ast);
static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req);
static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
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static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_request *req, char *rand, int reliable, char *header);
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static int transmit_request(struct sip_pvt *p, char *msg, int inc, int reliable, int newbranch);
static int transmit_request_with_auth(struct sip_pvt *p, char *msg, int inc, int reliable, int newbranch);
static int transmit_invite(struct sip_pvt *p, char *msg, int sendsdp, char *auth, char *authheader, char *vxml_url,char *distinctive_ring, char *osptoken,int init);
static int transmit_reinvite_with_sdp(struct sip_pvt *p);
static int transmit_info_with_digit(struct sip_pvt *p, char digit);
static int transmit_message_with_text(struct sip_pvt *p, char *text);
static int transmit_refer(struct sip_pvt *p, char *dest);
static struct sip_peer *temp_peer(char *name);
static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, char *msg, int init);
static void free_old_route(struct sip_route *route);
static int build_reply_digest(struct sip_pvt *p, char *orig_header, char *digest, int digest_len);
static int update_user_counter(struct sip_pvt *fup, int event);
static int sip_do_reload(void);
/*--- sip_debug_test_addr: See if we pass debug IP filter */
static inline int sip_debug_test_addr(struct sockaddr_in *addr)
{
if (sipdebug == 0)
return 0;
if (debugaddr.sin_addr.s_addr) {
if (((ntohs(debugaddr.sin_port) != 0)
&& (debugaddr.sin_port != addr->sin_port))
|| (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
return 0;
}
return 1;
}
static inline int sip_debug_test_pvt(struct sip_pvt *p)
{
return sip_debug_test_addr(((p->nat & SIP_NAT_ROUTE) ? &p->recv : &p->sa));
/*--- __sip_xmit: Transmit SIP message ---*/
static int __sip_xmit(struct sip_pvt *p, char *data, int len)
{
int res;
char iabuf[INET_ADDRSTRLEN];
if (p->nat & SIP_NAT_ROUTE)
res=sendto(sipsock, data, len, 0, (struct sockaddr *)&p->recv, sizeof(struct sockaddr_in));
else
res=sendto(sipsock, data, len, 0, (struct sockaddr *)&p->sa, sizeof(struct sockaddr_in));
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ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s returned %d: %s\n", data, len, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), res, strerror(errno));
static void sip_destroy(struct sip_pvt *p);
/*--- ast_sip_ouraddrfor: NAT fix - decide which IP address to use for ASterisk server? ---*/
/* Only used for outbound registrations */
static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us)
{
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* Using the localaddr structure built up with localnet statements
* apply it to their address to see if we need to substitute our
* externip or can get away with our internal bindaddr
*/
struct sockaddr_in theirs;
theirs.sin_addr = *them;
if (localaddr && externip.sin_addr.s_addr &&
ast_apply_ha(localaddr, &theirs)) {
char iabuf[INET_ADDRSTRLEN];
memcpy(us, &externip.sin_addr, sizeof(struct in_addr));
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ast_inet_ntoa(iabuf, sizeof(iabuf), *(struct in_addr *)&them->s_addr);
ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n", iabuf);
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}
else if (bindaddr.sin_addr.s_addr)
memcpy(us, &bindaddr.sin_addr, sizeof(struct in_addr));
else
return ast_ouraddrfor(them, us);
return 0;
}
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static int append_history(struct sip_pvt *p, char *event, char *data)
{
struct sip_history *hist, *prev;
char *c;
if (!recordhistory)
return 0;
hist = malloc(sizeof(struct sip_history));
if (hist) {
memset(hist, 0, sizeof(struct sip_history));
snprintf(hist->event, sizeof(hist->event), "%-15s %s", event, data);
/* Trim up nicely */
c = hist->event;
while(*c) {
if ((*c == '\r') || (*c == '\n')) {
*c = '\0';
break;
}
c++;
}
/* Enqueue into history */
prev = p->history;
if (prev) {
while(prev->next)
prev = prev->next;
prev->next = hist;
} else {
p->history = hist;
}
}
return 0;
}
/*--- retrans_pkt: Retransmit SIP message if no answer ---*/
struct sip_pkt *pkt=data, *prev, *cur;
char iabuf[INET_ADDRSTRLEN];
ast_mutex_lock(&pkt->owner->lock);
if (pkt->retrans < MAX_RETRANS) {
pkt->retrans++;
if (sip_debug_test_pvt(pkt->owner)) {
if (pkt->owner->nat & SIP_NAT_ROUTE)
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ast_verbose("Retransmitting #%d (NAT):\n%s\n to %s:%d\n", pkt->retrans, pkt->data, ast_inet_ntoa(iabuf, sizeof(iabuf), pkt->owner->recv.sin_addr), ntohs(pkt->owner->recv.sin_port));
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ast_verbose("Retransmitting #%d (no NAT):\n%s\n to %s:%d\n", pkt->retrans, pkt->data, ast_inet_ntoa(iabuf, sizeof(iabuf), pkt->owner->sa.sin_addr), ntohs(pkt->owner->sa.sin_port));
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append_history(pkt->owner, "ReTx", pkt->data);
__sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
res = 1;
ast_log(LOG_WARNING, "Maximum retries exceeded on call %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (pkt->flags & FLAG_FATAL) ? "Critical" : "Non-critical", (pkt->flags & FLAG_RESPONSE) ? "Response" : "Request");
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append_history(pkt->owner, "MaxRetries", pkt->flags & FLAG_FATAL ? "(Critical)" : "(Non-critical)");
if (pkt->flags & FLAG_FATAL) {
while(pkt->owner->owner && ast_mutex_trylock(&pkt->owner->owner->lock)) {
ast_mutex_unlock(&pkt->owner->lock);
usleep(1);
ast_mutex_lock(&pkt->owner->lock);
}
if (pkt->owner->owner) {
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ast_queue_hangup(pkt->owner->owner);
ast_mutex_unlock(&pkt->owner->owner->lock);
} else {
/* If no owner, destroy now */
pkt->owner->needdestroy = 1;
}
/* In any case, go ahead and remove the packet */
prev = NULL;
cur = pkt->owner->packets;
while(cur) {
if (cur == pkt)
break;
prev = cur;
cur = cur->next;
}
if (cur) {
if (prev)
prev->next = cur->next;
else
pkt->owner->packets = cur->next;
ast_mutex_unlock(&pkt->owner->lock);
free(cur);
pkt = NULL;
} else
ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
ast_mutex_unlock(&pkt->owner->lock);
/*--- __sip_reliable_xmit: transmit packet with retransmits ---*/
static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal)
pkt = malloc(sizeof(struct sip_pkt) + len + 1);
if (!pkt)
return -1;
memset(pkt, 0, sizeof(struct sip_pkt));
memcpy(pkt->data, data, len);
pkt->packetlen = len;
pkt->next = p->packets;
pkt->owner = p;
pkt->seqno = seqno;
if (fatal)
pkt->flags |= FLAG_FATAL;
pkt->retransid = ast_sched_add(sched, DEFAULT_RETRANS, retrans_pkt, pkt);
pkt->next = p->packets;
p->packets = pkt;
__sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
if (!strncasecmp(pkt->data, "INVITE", 6)) {
/* Note this is a pending invite */
p->pendinginvite = seqno;
}
/*--- __sip_autodestruct: Kill a call (called by scheduler) ---*/
static int __sip_autodestruct(void *data)
{
struct sip_pvt *p = data;
p->autokillid = -1;
ast_log(LOG_DEBUG, "Auto destroying call '%s'\n", p->callid);
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append_history(p, "AutoDestroy", "");
if (p->owner) {
ast_log(LOG_WARNING, "Autodestruct on call '%s' with owner in place\n", p->callid);
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ast_queue_hangup(p->owner);
} else {
sip_destroy(p);
}
return 0;
}
/*--- sip_scheddestroy: Schedule destruction of SIP call ---*/
static int sip_scheddestroy(struct sip_pvt *p, int ms)
{
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char tmp[80];
if (sip_debug_test_pvt(p))
ast_verbose("Scheduling destruction of call '%s' in %d ms\n", p->callid, ms);
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if (recordhistory) {
snprintf(tmp, sizeof(tmp), "%d ms", ms);
append_history(p, "SchedDestroy", tmp);
}
if (p->autokillid > -1)
ast_sched_del(sched, p->autokillid);
p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, p);
return 0;
}
/*--- sip_cancel_destroy: Cancel destruction of SIP call ---*/
static int sip_cancel_destroy(struct sip_pvt *p)
{
if (p->autokillid > -1)
ast_sched_del(sched, p->autokillid);
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append_history(p, "CancelDestroy", "");
/*--- __sip_ack: Acknowledges receipt of a packet and stops retransmission ---*/
static int __sip_ack(struct sip_pvt *p, int seqno, int resp, const char *msg)
{
struct sip_pkt *cur, *prev = NULL;
int res = -1;
int resetinvite = 0;
/* Just in case... */
if (!msg) msg = "___NEVER___";
if ((cur->seqno == seqno) && ((cur->flags & FLAG_RESPONSE) == resp) &&
((cur->flags & FLAG_RESPONSE) ||
(!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
if (!resp && (seqno == p->pendinginvite)) {
ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
p->pendinginvite = 0;
resetinvite = 1;
}
/* this is our baby */
if (prev)
prev->next = cur->next;
else
p->packets = cur->next;
if (cur->retransid > -1)
ast_sched_del(sched, cur->retransid);
free(cur);
res = 0;
break;
}
prev = cur;
cur = cur->next;
}
ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
/* Pretend to ack all packets */
static int __sip_pretend_ack(struct sip_pvt *p)
{
while(p->packets) {
__sip_ack(p, p->packets->seqno, (p->packets->flags & FLAG_RESPONSE), p->packets->data);
}
return 0;
}
/*--- __sip_semi_ack: Acks receipt of packet, keep it around (used for provisional responses) ---*/
static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, const char *msg)
struct sip_pkt *cur;
int res = -1;
cur = p->packets;
while(cur) {
if ((cur->seqno == seqno) && ((cur->flags & FLAG_RESPONSE) == resp) &&
((cur->flags & FLAG_RESPONSE) ||
(!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
/* this is our baby */
if (cur->retransid > -1)
ast_sched_del(sched, cur->retransid);
cur->retransid = -1;
res = 0;
break;
}
cur = cur->next;
}
ast_log(LOG_DEBUG, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
static void parse(struct sip_request *req);
static char *get_header(struct sip_request *req, char *name);
static void copy_request(struct sip_request *dst,struct sip_request *src);
static void parse_copy(struct sip_request *dst, struct sip_request *src)
{
memset(dst, 0, sizeof(*dst));
memcpy(dst->data, src->data, sizeof(dst->data));
dst->len = src->len;
parse(dst);
}
/*--- send_response: Transmit response on SIP request---*/
static int send_response(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno)
char iabuf[INET_ADDRSTRLEN];
if (sip_debug_test_pvt(p)) {
if (p->nat & SIP_NAT_ROUTE)
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ast_verbose("%sTransmitting (NAT):\n%s\n to %s:%d\n", reliable ? "Reliably " : "", req->data, ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port));
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ast_verbose("%sTransmitting (no NAT):\n%s\n to %s:%d\n", reliable ? "Reliably " : "", req->data, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port));
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if (reliable) {
if (recordhistory) {
parse_copy(&tmp, req);
snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
append_history(p, "TxRespRel", tmpmsg);
}
res = __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable > 1));
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} else {
if (recordhistory) {
parse_copy(&tmp, req);
snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
append_history(p, "TxResp", tmpmsg);
}
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}
if (res > 0)
res = 0;
return res;
}
/*--- send_request: Send SIP Request to the other part of the dialogue ---*/
static int send_request(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno)
char iabuf[INET_ADDRSTRLEN];
if (sip_debug_test_pvt(p)) {
if (p->nat & SIP_NAT_ROUTE)
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ast_verbose("%sTransmitting:\n%s (NAT) to %s:%d\n", reliable ? "Reliably " : "", req->data, ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port));
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ast_verbose("%sTransmitting:\n%s (no NAT) to %s:%d\n", reliable ? "Reliably " : "", req->data, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port));
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if (reliable) {
if (recordhistory) {
parse_copy(&tmp, req);
snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
append_history(p, "TxReqRel", tmpmsg);
}
res = __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1));
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} else {
if (recordhistory) {
parse_copy(&tmp, req);
snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
append_history(p, "TxReq", tmpmsg);
}
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}
/*--- url_decode: Decode SIP URL ---*/
static void url_decode(char *s)
{
char *o = s;
unsigned int tmp;
while(*s) {
switch(*s) {
case '%':
if (strlen(s) > 2) {
if (sscanf(s + 1, "%2x", &tmp) == 1) {
*o = tmp;
s += 2; /* Will be incremented once more when we break out */
break;
}
}
/* Fall through if something wasn't right with the formatting */
default:
*o = *s;
}
s++;
o++;
}
*o = '\0';
}
/*--- ditch_braces: Pick out text in braces from character string ---*/
static char *ditch_braces(char *tmp)
{
char *c = tmp;
char *n;
char *q;
if ((q = strchr(tmp, '"')) ) {
c = q + 1;
if ((q = strchr(c, '"')) )
c = q + 1;
else {
ast_log(LOG_WARNING, "No closing quote in '%s'\n", tmp);
c = tmp;
}
}
if ((n = strchr(c, '<')) ) {
c = n + 1;
while(*c && *c != '>') c++;
if (*c != '>') {
ast_log(LOG_WARNING, "No closing brace in '%s'\n", tmp);
} else {
*c = '\0';
}
return n+1;
}
return c;
}
/*--- sip_sendtext: Send SIP MESSAGE text within a call ---*/
/* Called from PBX core text message functions */
static int sip_sendtext(struct ast_channel *ast, char *text)
int debug=sip_debug_test_pvt(p);
if (debug)
ast_verbose("Sending text %s on %s\n", text, ast->name);
if (!p)
return -1;
if (!text || ast_strlen_zero(text))
ast_verbose("Really sending text %s on %s\n", text, ast->name);
transmit_message_with_text(p, text);
return 0;
static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, int expirey)
{
char port[10];
char ipaddr[20];