[asterisk-commits] oej: branch oej/codename-pineapple r44898 - in /team/oej/codename-pineapple/c...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Wed Oct 11 11:05:24 MST 2006


Author: oej
Date: Wed Oct 11 13:05:23 2006
New Revision: 44898

URL: http://svn.digium.com/view/asterisk?rev=44898&view=rev
Log:
Updates. Poking around. Having fun.

Added:
    team/oej/codename-pineapple/channels/sip3/sip3_network.c   (with props)
    team/oej/codename-pineapple/channels/sip3/sip3_refer.c   (with props)
    team/oej/codename-pineapple/channels/sip3/sip3_subscribe.c   (with props)
Modified:
    team/oej/codename-pineapple/channels/Makefile
    team/oej/codename-pineapple/channels/chan_sip3.c
    team/oej/codename-pineapple/channels/sip3/Makefile
    team/oej/codename-pineapple/channels/sip3/sip3.h

Modified: team/oej/codename-pineapple/channels/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/Makefile?rev=44898&r1=44897&r2=44898&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/Makefile (original)
+++ team/oej/codename-pineapple/channels/Makefile Wed Oct 11 13:05:23 2006
@@ -55,7 +55,7 @@
   LOADABLE_MODS:=
 endif
 
-all: _all
+all: _all sip3
 
 include $(ASTTOPDIR)/Makefile.moddir_rules
 
@@ -114,9 +114,9 @@
 
 $(if $(filter chan_misdn,$(EMBEDDED_MODS)),modules.link,chan_misdn.so): chan_misdn.o misdn_config.o misdn/isdn_lib.o misdn/isdn_msg_parser.o
 
-sip3/sip3_test.o: sip3/sip3_test.c
+sip3:	
 	$(MAKE) -C sip3 
 
-chan_sip3.so: chan_sip3.o sip3/sip3_test.o
+chan_sip3.so: chan_sip3.o sip3/sip3_network.o sip3/sip3_subscribe.o sip3/sip3_refer.o
 	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
 	$(CMD_PREFIX) $(CXX) $(SOLINK) -o $@ $< 

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=44898&r1=44897&r2=44898&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Wed Oct 11 13:05:23 2006
@@ -127,6 +127,7 @@
 	- Prove transaction engine by implementing PRACK
 	- Implement netsock API in this channel
 	- Add File's multithreading code
+	- Make debugaddr a ha list instead of one address and move it out of sipnet
 
 	- ... And much more
 */
@@ -191,10 +192,12 @@
 #include "asterisk/abstract_jb.h"
 #include "asterisk/compiler.h"
 #include "sip3/sip3.h"
+#include "sip3/sip3funcs.h"
 
 #define SIPLABEL	sip3
 
 
+/*! \brief various expiry times for registrations */
 static struct expiry_times expiry = {
 	.min_expiry = DEFAULT_MIN_EXPIRY,        /*!< Minimum accepted registration time */
 	.max_expiry = DEFAULT_MAX_EXPIRY,        /*!< Maximum accepted registration time */
@@ -202,8 +205,6 @@
 	.expiry = DEFAULT_EXPIRY,		/*!< Is this ever used ??? */
 };
 
-#define CALLERID_UNKNOWN        "Unknown"
-
 /*! \brief Global jitterbuffer configuration - by default, jb is disabled */
 static struct ast_jb_conf default_jbconf =
 {
@@ -212,158 +213,24 @@
 	.resync_threshold = -1,
 	.impl = ""
 };
-static struct ast_jb_conf global_jbconf;
 
 static const char config[] = "sip3.conf";
 static const char notify_config[] = "sip3_notify.conf";
 static int usecnt = 0;
 
-
-/*! \brief List of subscription event types for SUBSCRIBE requests */
-static const struct cfsubscription_types subscription_types[] = {
-	{ NONE,		   "-",        "unknown",	             "unknown" },
- 	/* RFC 4235: SIP Dialog event package */
-	{ DIALOG_INFO_XML, "dialog",   "application/dialog-info+xml", "dialog-info+xml" },
-	{ CPIM_PIDF_XML,   "presence", "application/cpim-pidf+xml",   "cpim-pidf+xml" },  /* RFC 3863 */
-	{ PIDF_XML,        "presence", "application/pidf+xml",        "pidf+xml" },       /* RFC 3863 */
-	{ XPIDF_XML,       "presence", "application/xpidf+xml",       "xpidf+xml" },       /* Pre-RFC 3863 with MS additions */
-	{ MWI_NOTIFICATION,	"message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
-};
-
-
-/*! XXX Note that sip_methods[i].id == i must hold or the code breaks */
-static const struct cfsip_methods sip_methods[] = {
-	{ SIP_UNKNOWN,	 RTP,    "-UNKNOWN-" },
-	{ SIP_RESPONSE,	 NO_RTP, "SIP/2.0" },
-	{ SIP_REGISTER,	 NO_RTP, "REGISTER" },
- 	{ SIP_OPTIONS,	 NO_RTP, "OPTIONS" },
-	{ SIP_NOTIFY,	 NO_RTP, "NOTIFY" },
-	{ SIP_INVITE,	 RTP,    "INVITE" },
-	{ SIP_ACK,	 NO_RTP, "ACK" },
-	{ SIP_PRACK,	 NO_RTP, "PRACK" },
-	{ SIP_BYE,	 NO_RTP, "BYE" },
-	{ SIP_REFER,	 NO_RTP, "REFER" },
-	{ SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE" },
-	{ SIP_MESSAGE,	 NO_RTP, "MESSAGE" },
-	{ SIP_UPDATE,	 NO_RTP, "UPDATE" },
-	{ SIP_INFO,	 NO_RTP, "INFO" },
-	{ SIP_CANCEL,	 NO_RTP, "CANCEL" },
-	{ SIP_PUBLISH,	 NO_RTP, "PUBLISH" }
-};
-
-/*! \brief List of well-known SIP options. If we get this in a require,
-   we should check the list and answer accordingly. */
-static const struct cfsip_options sip_options[] = {	/* XXX used in 3 places */
-	/* RFC3891: Replaces: header for transfer */
-	{ SIP_OPT_REPLACES,	SUPPORTED,	"replaces" },	
-	/* One version of Polycom firmware has the wrong label */
-	{ SIP_OPT_REPLACES,	SUPPORTED,	"replace" },	
-	/* RFC3262: PRACK 100% reliability */
-	{ SIP_OPT_100REL,	NOT_SUPPORTED,	"100rel" },	
-	/* RFC4028: SIP Session Timers */
-	{ SIP_OPT_TIMER,	NOT_SUPPORTED,	"timer" },
-	/* RFC3959: SIP Early session support */
-	{ SIP_OPT_EARLY_SESSION, NOT_SUPPORTED,	"early-session" },
-	/* RFC3911: SIP Join header support */
-	{ SIP_OPT_JOIN,		NOT_SUPPORTED,	"join" },
-	/* RFC3327: Path support */
-	{ SIP_OPT_PATH,		NOT_SUPPORTED,	"path" },
-	/* RFC3840: Callee preferences */
-	{ SIP_OPT_PREF,		NOT_SUPPORTED,	"pref" },
-	/* RFC3312: Precondition support */
-	{ SIP_OPT_PRECONDITION,	NOT_SUPPORTED,	"precondition" },
-	/* RFC3323: Privacy with proxies*/
-	{ SIP_OPT_PRIVACY,	NOT_SUPPORTED,	"privacy" },
-	/* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
-	{ SIP_OPT_SDP_ANAT,	NOT_SUPPORTED,	"sdp-anat" },
-	/* RFC3329: Security agreement mechanism */
-	{ SIP_OPT_SEC_AGREE,	NOT_SUPPORTED,	"sec_agree" },
-	/* SIMPLE events:  draft-ietf-simple-event-list-07.txt */
-	{ SIP_OPT_EVENTLIST,	NOT_SUPPORTED,	"eventlist" },
-	/* GRUU: Globally Routable User Agent URI's */
-	{ SIP_OPT_GRUU,		NOT_SUPPORTED,	"gruu" },
-	/* Target-dialog: draft-ietf-sip-target-dialog-03.txt */
-	{ SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED,	"tdialog" },
-	/* Disable the REFER subscription, RFC 4488 */
-	{ SIP_OPT_NOREFERSUB,	NOT_SUPPORTED,	"norefersub" },
-	/* ietf-sip-history-info-06.txt */
-	{ SIP_OPT_HISTINFO,	NOT_SUPPORTED,	"histinfo" },
-	/* ietf-sip-resource-priority-10.txt */
-	{ SIP_OPT_RESPRIORITY,	NOT_SUPPORTED,	"resource-priority" },
-};
-
-
 /* Default setttings are used as a channel setting and as a default when
    configuring devices */
-static int default_qualifycheck_ok;	/*!< Default qualify time when status is ok */
-static int default_qualifycheck_notok;	/*!< Default qualify time when statusis not ok */
-static int default_qualify;		/*!< Default Qualify= setting */
-static char default_context[AST_MAX_CONTEXT];
-static char default_subscribecontext[AST_MAX_CONTEXT];
-static char default_language[MAX_LANGUAGE];
-static char default_callerid[AST_MAX_EXTENSION];
-static char default_fromdomain[AST_MAX_EXTENSION];
-static char default_notifymime[AST_MAX_EXTENSION];
-static char default_vmexten[AST_MAX_EXTENSION];
-static char default_mohinterpret[MAX_MUSICCLASS];  /*!< Global setting for moh class to use when put on hold */
-static char default_mohsuggest[MAX_MUSICCLASS];	   /*!< Global setting for moh class to suggest when putting 
-                                                    *   a bridged channel on hold */
-static int default_maxcallbitrate;	/*!< Maximum bitrate for call */
-static struct ast_codec_pref default_prefs;		/*!< Default codec prefs */
-
 /* Global settings only apply to the channel */
-static int global_rtautoclear;
-static int global_notifyringing;	/*!< Send notifications on ringing */
-static int global_alwaysauthreject;	/*!< Send 401 Unauthorized for all failing requests */
-static int srvlookup;			/*!< SRV Lookup on or off. Default is off, RFC behavior is on */
-static int autocreatepeer;		/*!< Auto creation of peers at registration? Default off. */
-static int global_relaxdtmf;			/*!< Relax DTMF */
-static int global_rtptimeout;		/*!< Time out call if no RTP */
-static int global_rtpholdtimeout;
-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_allowsubscribe;	/*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE 
-					    the global setting is in globals_flags[1] */
-static int global_mwitime;		/*!< Time between MWI checks for peers */
-static unsigned int global_tos_sip;		/*!< IP type of service for SIP packets */
-static unsigned int global_tos_audio;		/*!< IP type of service for audio RTP packets */
-static unsigned int global_tos_video;		/*!< IP type of service for video RTP packets */
-static unsigned int global_tos_presence;	/*!< IP type of service for SIP presence packets */
-static int compactheaders;		/*!< send compact sip headers */
-static int recordhistory;		/*!< Record SIP history. Off by default */
-static int dumphistory;			/*!< Dump history to verbose before destroying SIP dialog */
-static char global_realm[MAXHOSTNAMELEN]; 		/*!< Default realm */
-static char global_regcontext[AST_MAX_CONTEXT];		/*!< Context for auto-extensions */
-static char global_useragent[AST_MAX_EXTENSION];	/*!< Useragent for the SIP channel */
-static int allow_external_domains;	/*!< Accept calls to external SIP domains? */
-static int global_callevents;		/*!< Whether we send manager events or not */
-static int global_t1min;		/*!< T1 roundtrip time minimum */
-static enum transfermodes global_allowtransfer;	/*!< SIP Refer restriction scheme */
-static int global_autoframing = 0;
-
-/*! \brief Codecs that we support by default: */
-static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
-static int noncodeccapability = AST_RTP_DTMF;
+static struct sip_globals global;
 
 /* Object counters */
-static int suserobjs = 0;                /*!< Static users */
-static int ruserobjs = 0;                /*!< Realtime users */
-static int speerobjs = 0;                /*!< Statis peers */
-static int rpeerobjs = 0;                /*!< Realtime peers */
-static int apeerobjs = 0;                /*!< Autocreated peer objects */
-static int regobjs = 0;                  /*!< Registry objects */
-
-static struct ast_flags global_flags[2] = {{0}};        /*!< global SIP_ flags */
-
+static struct channel_counters sipcounters = { 0, 0, 0, 0, 0, 0 };
 
 /*! \brief Protect the SIP dialog list (of sip_pvt's) */
 AST_MUTEX_DEFINE_STATIC(iflock);
 
 /*! \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(netlock);
 
 AST_MUTEX_DEFINE_STATIC(monlock);
 
@@ -378,36 +245,22 @@
 
 static struct sched_context *sched;     /*!< The scheduling context */
 static struct io_context *io;           /*!< The IO context */
-static int *sipsock_read_id;            /*!< ID of IO entry for sipsock FD */
 
 static AST_LIST_HEAD_STATIC(domain_list, domain);	/*!< The SIP domain list */
 AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
 
-/*!< This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
-static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
-
-static const struct c_referstatusstring {
-	enum referstatus status;
-	char *text;
-} referstatusstrings[] = {
-	{ REFER_IDLE,		"<none>" },
-	{ REFER_SENT,		"Request sent" },
-	{ REFER_RECEIVED,	"Request received" },
-	{ REFER_ACCEPTED,	"Accepted" },
-	{ REFER_RINGING,	"Target ringing" },
-	{ REFER_200OK,		"Done" },
-	{ REFER_FAILED,		"Failed" },
-	{ REFER_NOAUTH,		"Failed - auth failure" }
-} ;
 
 /* --- Linked lists of various objects --------*/
 
-/*! \brief  The user list: Users and friends */
+/*! \brief List of concurrent SIP dialogs */
+static struct sip_pvt *iflist = NULL;
+
+/*! \brief The user list: Users and friends */
 static struct ast_user_list {
 	ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer);
 } userl;
 
-/*! \brief  The peer list: Peers and Friends */
+/*! \brief The peer list: Peers and Friends */
 static struct ast_peer_list {
 	ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer);
 } peerl;
@@ -421,18 +274,8 @@
 /*! \todo Move the sip_auth list to AST_LIST */
 static struct sip_auth *authl = NULL;		/*!< Authentication list for realm authentication */
 
-/* --- Sockets and networking --------------*/
-static int sipsock  = -1;			/*!< Main socket for SIP network communication */
-static struct sockaddr_in bindaddr = { 0, };	/*!< The address we bind to */
-static struct sockaddr_in externip;		/*!< External IP address if we are behind NAT */
-static char externhost[MAXHOSTNAMELEN];		/*!< External host name (possibly with dynamic DNS and DHCP */
-static time_t externexpire = 0;			/*!< Expiration counter for re-resolving external host name in dynamic DNS */
-static int externrefresh = 10;
-static struct ast_ha *localaddr;		/*!< List of local networks, on the same side of NAT as this Asterisk */
-static struct in_addr __ourip;
-static struct sockaddr_in outboundproxyip;
-static int ourport;
-static struct sockaddr_in debugaddr;
+/* This structure needs to be encapsulated */
+extern struct sip_network sipnet;                      /*!< Sockets and networking */
 
 static struct ast_config *notify_types;		/*!< The list of manual NOTIFY types we know how to send */
 
@@ -440,7 +283,7 @@
 /*! \note This is added to help splitting up chan_sip.c into several files
 	in coming releases */
 
-/*--- PBX interface functions */
+/*--- PBX interface functions  - stays in this source code file */
 static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
 static int sip_devicestate(void *data);
 static int sip_sendtext(struct ast_channel *ast, const char *text);
@@ -456,7 +299,6 @@
 static int sip_senddigit_end(struct ast_channel *ast, char digit);
 
 /*--- Transmitting responses and requests */
-static int sipsock_read(int *id, int fd, short events, void *ignore);
 static int __sip_xmit(struct sip_pvt *p, char *data, int len);
 static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod);
 static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
@@ -605,8 +447,6 @@
 static int sip_show_user(int fd, int argc, char *argv[]);
 static int sip_show_registry(int fd, int argc, char *argv[]);
 static int sip_show_settings(int fd, int argc, char *argv[]);
-static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
-static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
 static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions);
 static int sip_show_channels(int fd, int argc, char *argv[]);
 static int sip_show_subscriptions(int fd, int argc, char *argv[]);
@@ -689,14 +529,9 @@
 /*--- 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 struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
 static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
-static int find_sip_method(const char *msg);
-static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported);
 static void parse_request(struct sip_request *req);
 static const char *get_header(const struct sip_request *req, const char *name);
-static char *referstatus2str(enum referstatus rstatus) attribute_pure;
-static int method_match(enum sipmethod id, const char *name);
 static void parse_copy(struct sip_request *dst, const struct sip_request *src);
 static char *get_in_brackets(char *tmp);
 static const char *find_alias(const char *name, const char *_default);
@@ -784,6 +619,7 @@
 static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
 static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
 
+
 /*! \brief Definition of this channel for PBX channel registration */
 static const struct ast_channel_tech sip_tech = {
 	.type = "SIP",
@@ -824,19 +660,6 @@
 	set_udptl_peer: sip_set_udptl_peer,
 };
 
-/*! \brief Convert transfer status to string */
-static char *referstatus2str(enum referstatus rstatus)
-{
-	int i = (sizeof(referstatusstrings) / sizeof(referstatusstrings[0]));
-	int x;
-
-	for (x = 0; x < i; x++) {
-		if (referstatusstrings[x].status ==  rstatus)
-			return (char *) referstatusstrings[x].text;
-	}
-	return "";
-}
-
 /*! \brief Initialize the initital request packet in the pvt structure.
  	This packet is used for creating replies and future requests in
 	a dialog */
@@ -852,88 +675,15 @@
 		ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
 }
 
-
-/*! \brief returns true if 'name' (with optional trailing whitespace)
- * matches the sip method 'id'.
- * Strictly speaking, SIP methods are case SENSITIVE, but we do
- * a case-insensitive comparison to be more tolerant.
- * following Jon Postel's rule: Be gentle in what you accept, strict with what you send
- */
-static int method_match(enum sipmethod id, const char *name)
-{
-	int len = strlen(sip_methods[id].text);
-	int l_name = name ? strlen(name) : 0;
-	/* true if the string is long enough, and ends with whitespace, and matches */
-	return (l_name >= len && name[len] < 33 &&
-		!strncasecmp(sip_methods[id].text, name, len));
-}
-
-/*! \brief  find_sip_method: Find SIP method from header */
-static int find_sip_method(const char *msg)
-{
-	int i, res = 0;
-	
-	if (ast_strlen_zero(msg))
-		return 0;
-	for (i = 1; i < (sizeof(sip_methods) / sizeof(sip_methods[0])) && !res; i++) {
-		if (method_match(i, msg))
-			res = sip_methods[i].id;
-	}
-	return res;
-}
-
-/*! \brief Parse supported header in incoming packet */
-static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported)
-{
-	char *next, *sep;
-	char *temp = ast_strdupa(supported);
-	unsigned int profile = 0;
-	int i, found;
-
-	if (ast_strlen_zero(supported) )
-		return 0;
-
-	if (option_debug > 2 && sipdebug)
-		ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported);
-
-	for (next = temp; next; next = sep) {
-		found = FALSE;
-		if ( (sep = strchr(next, ',')) != NULL)
-			*sep++ = '\0';
-		next = ast_skip_blanks(next);
-		if (option_debug > 2 && sipdebug)
-			ast_log(LOG_DEBUG, "Found SIP option: -%s-\n", next);
-		for (i=0; i < (sizeof(sip_options) / sizeof(sip_options[0])); i++) {
-			if (!strcasecmp(next, sip_options[i].text)) {
-				profile |= sip_options[i].id;
-				found = TRUE;
-				if (option_debug > 2 && sipdebug)
-					ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next);
-				break;
-			}
-		}
-		if (!found && option_debug > 2 && sipdebug) {
-			if (!strncasecmp(next, "x-", 2))
-				ast_log(LOG_DEBUG, "Found private SIP option, not supported: %s\n", next);
-			else
-				ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next);
-		}
-	}
-
-	if (pvt)
-		pvt->sipoptions = profile;
-	return profile;
-}
-
 /*! \brief See if we pass debug IP filter */
 static inline int sip_debug_test_addr(const struct sockaddr_in *addr) 
 {
 	if (!sipdebug)
 		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))
+	if (sipnet.debugaddr.sin_addr.s_addr) {
+		if (((ntohs(sipnet.debugaddr.sin_port) != 0)
+			&& (sipnet.debugaddr.sin_port != addr->sin_port))
+			|| (sipnet.debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
 			return 0;
 	}
 	return 1;
@@ -964,7 +714,7 @@
 {
 	int res;
 	const struct sockaddr_in *dst = sip_real_dst(p);
-	res=sendto(sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
+	res=sendto(sipnet.sipsock, data, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
 
 	if (res != len)
 		ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));
@@ -980,7 +730,7 @@
 
 	/* z9hG4bK is a magic cookie.  See RFC 3261 section 8.1.1.7 */
 	ast_string_field_build(p, via, "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x%s",
-			 ast_inet_ntoa(p->ourip), ourport, p->branch, rport);
+			 ast_inet_ntoa(p->ourip), sipnet_ourport(), p->branch, rport);
 }
 
 /*! \brief NAT fix - decide which IP address to use for ASterisk server?
@@ -998,32 +748,28 @@
 	theirs.sin_addr = *them;
 	ours.sin_addr = *us;
 
-	if (localaddr && externip.sin_addr.s_addr &&
-	    ast_apply_ha(localaddr, &theirs) &&
-	    !ast_apply_ha(localaddr, &ours)) {
-		if (externexpire && time(NULL) >= externexpire) {
+	if (sipnet.localaddr && sipnet.externip.sin_addr.s_addr &&
+	    ast_apply_ha(sipnet.localaddr, &theirs) &&
+	    !ast_apply_ha(sipnet.localaddr, &ours)) {
+		if (sipnet.externexpire && time(NULL) >= sipnet.externexpire) {
 			struct ast_hostent ahp;
 			struct hostent *hp;
 
-			externexpire = time(NULL) + externrefresh;
-			if ((hp = ast_gethostbyname(externhost, &ahp))) {
-				memcpy(&externip.sin_addr, hp->h_addr, sizeof(externip.sin_addr));
+			sipnet.externexpire = time(NULL) + sipnet.externrefresh;
+			if ((hp = ast_gethostbyname(sipnet.externhost, &ahp))) {
+				memcpy(&sipnet.externip.sin_addr, hp->h_addr, sizeof(sipnet.externip.sin_addr));
 			} else
-				ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
-		}
-		*us = externip.sin_addr;
+				ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", sipnet.externhost);
+		}
+		*us = sipnet.externip.sin_addr;
 		if (option_debug) {
 			ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n", 
 				ast_inet_ntoa(*(struct in_addr *)&them->s_addr));
 		}
-	} else if (bindaddr.sin_addr.s_addr)
-		*us = bindaddr.sin_addr;
+	} else if (sipnet.bindaddr.sin_addr.s_addr)
+		*us = sipnet.bindaddr.sin_addr;
 	return AST_SUCCESS;
 }
-
-/*! \brief Append to SIP dialog history 
-	\return Always returns 0 */
-#define append_history(p, event, fmt , args... )	append_history_full(p, "%-15s " fmt, event, ## args)
 
 static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
 	__attribute__ ((format (printf, 2, 3)));
@@ -1053,7 +799,7 @@
 {
 	va_list ap;
 
-	if (!recordhistory || !p)
+	if (!global.recordhistory || !p)
 		return;
 	va_start(ap, fmt);
 	append_history_va(p, fmt, ap);
@@ -1075,12 +821,12 @@
 		pkt->retrans++;
  		if (!pkt->timer_t1) {	/* Re-schedule using timer_a and timer_t1 */
 			if (sipdebug && option_debug > 3)
- 				ast_log(LOG_DEBUG, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_methods[pkt->method].text, pkt->method);
+ 				ast_log(LOG_DEBUG, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_method2txt(pkt->method), pkt->method);
 		} else {
  			int siptimer_a;
 
  			if (sipdebug && option_debug > 3)
- 				ast_log(LOG_DEBUG, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method);
+ 				ast_log(LOG_DEBUG, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_method2txt(pkt->method), pkt->method);
  			if (!pkt->timer_a)
  				pkt->timer_a = 2 ;
  			else
@@ -1220,7 +966,7 @@
 		ast_log(LOG_DEBUG, "Auto destroying SIP dialog '%s'\n", p->callid);
 	append_history(p, "AutoDestroy", "%s", p->callid);
 	if (p->owner) {
-		ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
+		ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_method2txt(p->method));
 		ast_queue_hangup(p->owner);
 	} else if (p->refer) {
 		transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
@@ -1239,8 +985,8 @@
 		ms = p->timer_t1 * 64;
 	}
 	if (sip_debug_test_pvt(p))
-		ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
-	if (recordhistory)
+		ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_method2txt(p->method));
+	if (global.recordhistory)
 		append_history(p, "SchedDestroy", "%d ms", ms);
 
 	if (p->autokillid > -1)
@@ -1267,7 +1013,7 @@
 	char *msg;
 	int res = FALSE;
 
-	msg = sip_methods[sipmethod].text;
+	msg = sip_method2txt(sipmethod);
 
 	ast_mutex_lock(&p->lock);
 	for (cur = p->packets; cur; prev = cur, cur = cur->next) {
@@ -1306,7 +1052,7 @@
 	while (p->packets) {
 		int method;
 		if (cur == p->packets) {
-			ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
+			ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_method2txt(cur->method));
 			return;
 		}
 		cur = p->packets;
@@ -1327,7 +1073,7 @@
 			/* this is our baby */
 			if (cur->retransid > -1) {
 				if (option_debug > 3 && sipdebug)
-					ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
+					ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_method2txt(sipmethod));
 				ast_sched_del(sched, cur->retransid);
 			}
 			cur->retransid = -1;
@@ -1374,11 +1120,11 @@
 			ast_inet_ntoa(dst->sin_addr),
 			ntohs(dst->sin_port), req->data);
 	}
-	if (recordhistory) {
+	if (global.recordhistory) {
 		struct sip_request tmp;
 		parse_copy(&tmp, req);
 		append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), 
-			(tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? tmp.rlPart2 : sip_methods[tmp.method].text);
+			(tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? tmp.rlPart2 : sip_method2txt(tmp.method));
 	}
 	res = (reliable) ?
 		__sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
@@ -1400,10 +1146,10 @@
 		else
 			ast_verbose("%sTransmitting (no NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->sa.sin_addr), ntohs(p->sa.sin_port), req->data);
 	}
-	if (recordhistory) {
+	if (global.recordhistory) {
 		struct sip_request tmp;
 		parse_copy(&tmp, req);
-		append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
+		append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_method2txt(tmp.method));
 	}
 	res = (reliable) ?
 		__sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1), req->method) :
@@ -1496,7 +1242,7 @@
 	
 	if (ast_strlen_zero(sysname))	/* No system name, disable this */
 		sysname = NULL;
-	else if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTSAVE_SYSNAME))
+	else if (ast_test_flag(&global.flags[1], SIP_PAGE2_RTSAVE_SYSNAME))
 		syslabel = "regserver";
 
 	if (fc)
@@ -1515,11 +1261,11 @@
 	char multi[256];
 	char *stringp, *ext, *context;
 
-	/* XXX note that global_regcontext is both a global 'enable' flag and
+	/* XXX note that global.regcontext is both a global 'enable' flag and
 	 * the name of the global regexten context, if not specified
 	 * individually.
 	 */
-	if (ast_strlen_zero(global_regcontext))
+	if (ast_strlen_zero(global.regcontext))
 		return;
 
 	ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi));
@@ -1532,7 +1278,7 @@
 				continue;
 			}
 		} else {
-			context = global_regcontext;
+			context = global.regcontext;
 		}
 		if (onoff)
 			ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop",
@@ -1570,17 +1316,17 @@
 		clear_realm_authentication(device->auth);
 		device->auth = (struct sip_auth *) NULL;
 		if (ast_test_flag((&device->flags[1]), SIP_PAGE2_SELFDESTRUCT))
-			apeerobjs--;
+			sipcounters.autocreated_peers--;
 		else if (ast_test_flag(&device->flags[0], SIP_REALTIME))
-			rpeerobjs--;
+			sipcounters.realtime_peers--;
 		else
-			speerobjs--;
+			sipcounters.static_peers--;
 	} 
 	if (device->type & SIP_USER) {	/* SIP_USER */
 		if (ast_test_flag(&device->flags[0], SIP_REALTIME))
-			ruserobjs--;
+			sipcounters.realtime_users--;
 		else
-			suserobjs--;
+			sipcounters.static_users--;
 	}
 	if (device->dnsmgr)
 		ast_dnsmgr_release(device->dnsmgr);
@@ -1591,7 +1337,7 @@
 static void update_peer(struct sip_peer *p, int expiry)
 {
 	int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
-	if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTUPDATE) &&
+	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);
 	}
@@ -1641,20 +1387,20 @@
 	}
 
 	/* Peer found in realtime, now build it in memory */
-	peer = build_peer(newpeername, var, NULL, !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS));
+	peer = build_peer(newpeername, var, NULL, !ast_test_flag(&global.flags[1], SIP_PAGE2_RTCACHEFRIENDS));
 	if (!peer) {
 		ast_variables_destroy(var);
 		return NULL;
 	}
 
-	if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
+	if (ast_test_flag(&global.flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
 		/* Cache peer */
-		ast_copy_flags(&peer->flags[1],&global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
-		if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
+		ast_copy_flags(&peer->flags[1],&global.flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
+		if (ast_test_flag(&global.flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
 			if (peer->expire > -1) {
 				ast_sched_del(sched, peer->expire);
 			}
-			peer->expire = ast_sched_add(sched, (global_rtautoclear) * 1000, expire_register, (void *)peer);
+			peer->expire = ast_sched_add(sched, (global.rtautoclear) * 1000, expire_register, (void *)peer);
 		}
 		ASTOBJ_CONTAINER_LINK(&peerl,peer);
 	} else {
@@ -1715,20 +1461,20 @@
 		}
 	}
 
-	user = build_user(username, var, !ast_test_flag((&global_flags[1]), SIP_PAGE2_RTCACHEFRIENDS)); 
+	user = build_user(username, var, !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)) {
+	if (ast_test_flag(&global.flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
 		ast_set_flag(&user->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
-		suserobjs++;
+		sipcounters.static_users++;
 		ASTOBJ_CONTAINER_LINK(&userl,user);
 	} else {
 		/* Move counter from s to r... */
-		suserobjs--;
-		ruserobjs++;
+		sipcounters.static_users--;
+		sipcounters.realtime_users++;
 		ast_set_flag(&user->flags[0], SIP_REALTIME);
 	}
 	ast_variables_destroy(var);
@@ -1770,7 +1516,7 @@
 	}
 	dialog->prefs = peer->prefs;
 	if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
-		dialog->t38.capability = global_t38_capability;
+		dialog->t38.capability = global.t38_capability;
 		if (dialog->udptl) {
 			if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_FEC )
 				dialog->t38.capability |= T38FAX_UDP_EC_FEC;
@@ -1842,7 +1588,7 @@
 	/* Set timer T1 to RTT for this peer (if known by qualify=) */
 	/* Minimum is settable or default to 100 ms */
 	if (peer->maxms && peer->lastms)
-		dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
+		dialog->timer_t1 = peer->lastms < global.t1min ? global.t1min : peer->lastms;
 	if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
 	    (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
 		dialog->noncodeccapability |= AST_RTP_DTMF;
@@ -1886,8 +1632,8 @@
 		return res;
 	}
 	hostn = peer;
-	portno = port ? atoi(port) : DEFAULT_SIP_PORT;
-	if (srvlookup) {
+	portno = port ? atoi(port) : STANDARD_SIP_PORT;
+	if (global.srvlookup) {
 		char service[MAXHOSTNAMELEN];
 		int tportno;
 		int ret;
@@ -2045,7 +1791,7 @@
 	if (reg->timeout > -1)
 		ast_sched_del(sched, reg->timeout);
 	ast_string_field_free_all(reg);
-	regobjs--;
+	sipcounters.registry_objects--;
 	free(reg);
 	
 }
@@ -2057,13 +1803,13 @@
 	struct sip_pkt *cp;
 
 	if (sip_debug_test_pvt(p) || option_debug > 2)
-		ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
+		ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_method2txt(p->method));
 
 	/* Remove link from peer to subscription of MWI */
 	if (p->relatedpeer && p->relatedpeer->mwipvt)
 		p->relatedpeer->mwipvt = NULL;
 
-	if (dumphistory)
+	if (global.dumphistory)
 		sip_dump_history(p);
 
 	if (p->options)
@@ -2535,7 +2281,7 @@
 				transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
 
 				/* Get RTCP quality before end of call */
-				if (recordhistory) {
+				if (global.recordhistory) {
 					if (p->rtp)
 						append_history(p, "RTCPaudio", "Quality:%s", audioqos);
 					if (p->vrtp)
@@ -2907,7 +2653,7 @@
 	else if (i->capability)		/* Our configured capability for this peer */
 		what = i->capability;
 	else
-		what = global_capability;	/* Global codec support */
+		what = global.capability;	/* Global codec support */
 
 	/* Set the native formats for audio  and merge in video */
 	tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
@@ -2953,7 +2699,7 @@
 	if (ast_test_flag(&i->flags[0], SIP_DTMF) ==  SIP_DTMF_INBAND) {
 		i->vad = ast_dsp_new();
 		ast_dsp_set_features(i->vad, DSP_FEATURE_DTMF_DETECT);
-		if (global_relaxdtmf)
+		if (global.relaxdtmf)
 			ast_dsp_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
 	}
 	if (i->rtp) {
@@ -3011,7 +2757,7 @@
 		pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
 	ast_setstate(tmp, state);
 	if (i->rtp)
-		ast_jb_configure(tmp, &global_jbconf);
+		ast_jb_configure(tmp, &global.jbconf);
 	if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
 		ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
 		tmp->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
@@ -3022,7 +2768,7 @@
 	for (v = i->chanvars ; v ; v = v->next)
 		pbx_builtin_setvar_helper(tmp,v->name,v->value);
 
-	if (recordhistory)
+	if (global.recordhistory)
 		append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
 
 	return tmp;
@@ -3311,7 +3057,7 @@
 	p->autokillid = -1;
 	p->subscribed = NONE;
 	p->stateid = -1;
-	p->prefs = default_prefs;		/* Set default codecs for this call */
+	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) */
@@ -3319,25 +3065,25 @@
 	if (sin) {
 		p->sa = *sin;
 		if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip))
-			p->ourip = __ourip;
+			p->ourip = sipnet.__ourip;
 	} else
-		p->ourip = __ourip;
+		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_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);
 
 	p->branch = ast_random();	
 	make_our_tag(p->tag, sizeof(p->tag));
 	p->ocseq = INITIAL_CSEQ;
 
-	if (sip_methods[intended_method].need_rtp) {
-		p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+	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, bindaddr.sin_addr);
+			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, bindaddr.sin_addr);
+			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));
@@ -3351,24 +3097,24 @@
 		}
 		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);
+		ast_rtp_settos(p->rtp, global.tos_audio);
 		if (p->vrtp) {
-			ast_rtp_settos(p->vrtp, global_tos_video);
+			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 = default_maxcallbitrate;
+			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) {
 		int natflags;
 		/* Setup NAT structure according to global settings if we have an address */
-		ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT);
+		ast_copy_flags(&p->flags[0], &global.flags[0], SIP_NAT);
 		p->recv = *sin;
 		natflags = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE;
 		if (p->rtp)
@@ -3380,22 +3126,22 @@
 	}
 
 	if (p->method != SIP_REGISTER)
-		ast_string_field_set(p, fromdomain, default_fromdomain);
+		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, default_mohinterpret);
-	ast_string_field_set(p, mohsuggest, default_mohsuggest);
-	p->capability = global_capability;
-	p->allowtransfer = global_allowtransfer;
+	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;
+		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)
@@ -3405,7 +3151,7 @@
 		p->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
 		p->t38.jointcapability = p->t38.capability;
 	}
-	ast_string_field_set(p, context, default_context);

[... 2807 lines stripped ...]


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