[asterisk-commits] oej: branch oej/codename-pineapple r44743 - in /team/oej/codename-pineapple/c...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Sat Oct 7 14:36:26 MST 2006


Author: oej
Date: Sat Oct  7 16:36:25 2006
New Revision: 44743

URL: http://svn.digium.com/view/asterisk?rev=44743&view=rev
Log:
Moving declarations to include file

Modified:
    team/oej/codename-pineapple/channels/chan_sip3.c
    team/oej/codename-pineapple/channels/sip3/sip3.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=44743&r1=44742&r2=44743&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Sat Oct  7 16:36:25 2006
@@ -197,25 +197,6 @@
 
 #define CALLERID_UNKNOWN        "Unknown"
 
-#define DEFAULT_QUALIFY_MAXMS                2000             /*!< Qualification: Must be faster than 2 seconds by default */
-#define DEFAULT_QUALIFY_FREQ_OK      60 * 1000        /*!< Qualification: How often to check for the host to be up */
-			
-#define DEFAULT_QUALIFY_FREQ_NOTOK   10 * 1000        /*!< Qualification: How often to check, if the host is down... */
-
-#define DEFAULT_RETRANS              1000             /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
-#define MAX_RETRANS                  6                /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
-#define SIP_TRANS_TIMEOUT            32000            /*!< SIP request timeout (rfc 3261) 64*T1 
-                                                      \todo Use known T1 for timeout (peerpoke)
-                                                      */
-#define DEFAULT_TRANS_TIMEOUT        -1               /* Use default SIP transaction timeout */
-#define MAX_AUTHTRIES                3                /*!< Try authentication three times, then fail */
-
-#define SIP_MAX_HEADERS              64               /*!< Max amount of SIP headers to read */
-#define SIP_MAX_LINES                64               /*!< Max amount of lines in SIP attachment (like SDP) */
-#define SIP_MAX_PACKET               4096             /*!< Also from RFC 3261 (2543), should sub headers tho */
-
-#define INITIAL_CSEQ                 101              /*!< our initial sip sequence number */
-
 /*! \brief Global jitterbuffer configuration - by default, jb is disabled */
 static struct ast_jb_conf default_jbconf =
 {
@@ -230,59 +211,6 @@
 static const char notify_config[] = "sip3_notify.conf";
 static int usecnt = 0;
 
-
-#define RTP 	1
-#define NO_RTP	0
-
-/*! \brief Authorization scheme for call transfers 
-\note Not a bitfield flag, since there are plans for other modes,
-	like "only allow transfers for authenticated devices" */
-enum transfermodes {
-	TRANSFER_OPENFORALL,            /*!< Allow all SIP transfers */
-	TRANSFER_CLOSED,                /*!< Allow no SIP transfers */
-};
-
-
-enum sip_result {
-	AST_SUCCESS = 0,
-	AST_FAILURE = -1,
-};
-
-/* Do _NOT_ make any changes to this enum, or the array following it;
-   if you think you are doing the right thing, you are probably
-   not doing the right thing. If you think there are changes
-   needed, get someone else to review them first _before_
-   submitting a patch. If these two lists do not match properly
-   bad things will happen.
-*/
-
-enum objecttype {
-	SIP_USER = (1 << 0),		/* USER places calls to the PBX */
-	SIP_PEER = (1 << 1),		/* Peer receives calls from PBX (and places calls) */
-};
-
-enum xmittype {
-	XMIT_CRITICAL = 2,              /*!< Transmit critical SIP message reliably, with re-transmits.
-                                              If it fails, it's critical and will cause a teardown of the session */
-	XMIT_RELIABLE = 1,              /*!< Transmit SIP message reliably, with re-transmits */
-	XMIT_UNRELIABLE = 0,            /*!< Transmit SIP message without bothering with re-transmits */
-};
-
-enum parse_register_result {
-	PARSE_REGISTER_FAILED,
-	PARSE_REGISTER_UPDATE,
-	PARSE_REGISTER_QUERY,
-};
-
-enum subscriptiontype { 
-	NONE = 0,
-	TIMEOUT,
-	XPIDF_XML,
-	DIALOG_INFO_XML,
-	CPIM_PIDF_XML,
-	PIDF_XML,
-	MWI_NOTIFICATION
-};
 
 static const struct cfsubscription_types {
 	enum subscriptiontype type;
@@ -297,59 +225,6 @@
 	{ 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 */
-};
-
-/*! \brief SIP Request methods known by Asterisk */
-enum sipmethod {
-	SIP_UNKNOWN,		/* Unknown response */
-	SIP_RESPONSE,		/* Not request, response to outbound request */
-	SIP_REGISTER,
-	SIP_OPTIONS,
-	SIP_NOTIFY,
-	SIP_INVITE,
-	SIP_ACK,
-	SIP_PRACK,		/* Not supported at all */
-	SIP_BYE,
-	SIP_REFER,
-	SIP_SUBSCRIBE,
-	SIP_MESSAGE,
-	SIP_UPDATE,		/* We can send UPDATE; but not accept it */
-	SIP_INFO,
-	SIP_CANCEL,
-	SIP_PUBLISH,		/* Not supported at all */
-};
-
-/*! \brief Authentication types - proxy or www authentication 
-	\note Endpoints, like Asterisk, should always use WWW authentication to
-	allow multiple authentications in the same call - to the proxy and
-	to the end point.
-*/
-enum sip_auth_type {
-	PROXY_AUTH,
-	WWW_AUTH,
-};
-
-/*! \brief Authentication result from check_auth* functions */
-enum check_auth_result {
-	AUTH_SUCCESSFUL = 0,
-	AUTH_CHALLENGE_SENT = 1,
-	AUTH_SECRET_FAILED = -1,
-	AUTH_USERNAME_MISMATCH = -2,
-	AUTH_NOT_FOUND = -3,
-	AUTH_FAKE_AUTH = -4,
-	AUTH_UNKNOWN_DOMAIN = -5,
-};
-
-/*! \brief States for outbound registrations (with register= lines in sip.conf */
-enum sipregistrystate {
-	REG_STATE_UNREGISTERED = 0,	/*!< We are not registred */
-	REG_STATE_REGSENT,	/*!< Registration request sent */
-	REG_STATE_AUTHSENT,	/*!< We have tried to authenticate */
-	REG_STATE_REGISTERED,	/*!< Registred and done */
-	REG_STATE_REJECTED,	/*!< Registration rejected */
-	REG_STATE_TIMEOUT,	/*!< Registration timed out */
-	REG_STATE_NOAUTH,	/*!< We have no accepted credentials */
-	REG_STATE_FAILED,	/*!< Registration failed after several tries */
 };
 
 
@@ -376,37 +251,6 @@
 	{ SIP_CANCEL,	 NO_RTP, "CANCEL" },
 	{ SIP_PUBLISH,	 NO_RTP, "PUBLISH" }
 };
-
-/*!  Define SIP option tags, used in Require: and Supported: headers 
- 	We need to be aware of these properties in the phones to use 
-	the replace: header. We should not do that without knowing
-	that the other end supports it... 
-	This is nothing we can configure, we learn by the dialog
-	Supported: header on the REGISTER (peer) or the INVITE
-	(other devices)
-	We are not using many of these today, but will in the future.
-	This is documented in RFC 3261
-*/
-#define SUPPORTED		1
-#define NOT_SUPPORTED		0
-
-#define SIP_OPT_REPLACES	(1 << 0)
-#define SIP_OPT_100REL		(1 << 1)
-#define SIP_OPT_TIMER		(1 << 2)
-#define SIP_OPT_EARLY_SESSION	(1 << 3)
-#define SIP_OPT_JOIN		(1 << 4)
-#define SIP_OPT_PATH		(1 << 5)
-#define SIP_OPT_PREF		(1 << 6)
-#define SIP_OPT_PRECONDITION	(1 << 7)
-#define SIP_OPT_PRIVACY		(1 << 8)
-#define SIP_OPT_SDP_ANAT	(1 << 9)
-#define SIP_OPT_SEC_AGREE	(1 << 10)
-#define SIP_OPT_EVENTLIST	(1 << 11)
-#define SIP_OPT_GRUU		(1 << 12)
-#define SIP_OPT_TARGET_DIALOG	(1 << 13)
-#define SIP_OPT_NOREFERSUB	(1 << 14)
-#define SIP_OPT_HISTINFO	(1 << 15)
-#define SIP_OPT_RESPRIORITY	(1 << 16)
 
 /*! \brief List of well-known SIP options. If we get this in a require,
    we should check the list and answer accordingly. */
@@ -454,45 +298,6 @@
 };
 
 
-/*! \brief SIP Methods we support */
-#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
-
-/*! \brief SIP Extensions we support */
-#define SUPPORTED_EXTENSIONS "replaces" 
-
-
-/* Default values, set and reset in reload_config before reading configuration */
-/* These are default values in the source. There are other recommended values in the
-   sip.conf.sample for new installations. These may differ to keep backwards compatibility,
-   yet encouraging new behaviour on new installations 
- */
-#define DEFAULT_SIP_PORT	5060	/*!< From RFC 3261 (former 2543) */
-#define DEFAULT_CONTEXT		"default"
-#define DEFAULT_MOHINTERPRET    "default"
-#define DEFAULT_MOHSUGGEST      ""
-#define DEFAULT_VMEXTEN 	"asterisk"
-#define DEFAULT_CALLERID 	"asterisk"
-#define DEFAULT_NOTIFYMIME 	"application/simple-message-summary"
-#define DEFAULT_MWITIME 	10
-#define DEFAULT_ALLOWGUEST	TRUE
-#define DEFAULT_SRVLOOKUP	FALSE		/*!< Recommended setting is ON */
-#define DEFAULT_COMPACTHEADERS	FALSE
-#define DEFAULT_TOS_SIP         0               /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_TOS_AUDIO       0               /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_TOS_VIDEO       0               /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_TOS_PRESENCE    0               /*!< Presence notifications does not need SIP priority */
-#define DEFAULT_ALLOW_EXT_DOM	TRUE
-#define DEFAULT_REALM		"asterisk"
-#define DEFAULT_NOTIFYRINGING	TRUE
-#define DEFAULT_AUTOCREATEPEER	FALSE
-#define DEFAULT_QUALIFY		FALSE
-#define DEFAULT_T1MIN		100		/*!< 100 MS for minimal roundtrip time */
-#define DEFAULT_MAX_CALL_BITRATE (384)		/*!< Max bitrate for video */
-#ifndef DEFAULT_USERAGENT
-#define DEFAULT_USERAGENT "Asterisk PBX"	/*!< Default Useragent: header unless re-defined in sip.conf */
-#endif
-
-
 /* 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 */
@@ -580,262 +385,11 @@
 static struct io_context *io;           /*!< The IO context */
 static int *sipsock_read_id;            /*!< ID of IO entry for sipsock FD */
 
-#define DEC_CALL_LIMIT	0
-#define INC_CALL_LIMIT	1
-#define DEC_CALL_RINGING 2
-#define INC_CALL_RINGING 3
-
-/*! \brief sip_request: The data grabbed from the UDP socket */
-struct sip_request {
-	char *rlPart1; 	        /*!< SIP Method Name or "SIP/2.0" protocol version */
-	char *rlPart2; 	        /*!< The Request URI or Response Status */
-	int len;                /*!< Length */
-	int headers;            /*!< # of SIP Headers */
-	int method;             /*!< Method of this request */
-	int lines;              /*!< Body Content */
-	unsigned int flags;     /*!< SIP_PKT Flags for this packet */
-	char *header[SIP_MAX_HEADERS];
-	char *line[SIP_MAX_LINES];
-	char data[SIP_MAX_PACKET];
-	unsigned int sdp_start; /*!< the line number where the SDP begins */
-	unsigned int sdp_end;   /*!< the line number where the SDP ends */
-};
-
-/*
- * A sip packet is stored into the data[] buffer, with the header followed
- * by an empty line and the body of the message.
- * On outgoing packets, data is accumulated in data[] with len reflecting
- * the next available byte, headers and lines count the number of lines
- * in both parts. There are no '\0' in data[0..len-1].
- *
- * On received packet, the input read from the socket is copied into data[],
- * len is set and the string is NUL-terminated. Then a parser fills up
- * the other fields -header[] and line[] to point to the lines of the
- * message, rlPart1 and rlPart2 parse the first lnie as below:
- *
- * Requests have in the first line	METHOD URI SIP/2.0
- *	rlPart1 = method; rlPart2 = uri;
- * Responses have in the first line	SIP/2.0 code description
- *	rlPart1 = SIP/2.0; rlPart2 = code + description;
- *
- */
-
-/*! \brief structure used in transfers */
-struct sip_dual {
-	struct ast_channel *chan1;	/*!< First channel involved */
-	struct ast_channel *chan2;	/*!< Second channel involved */
-	struct sip_request req;		/*!< Request that caused the transfer (REFER) */
-	int seqno;			/*!< Sequence number */
-};
-
-struct sip_pkt;
-
-/*! \brief Parameters to the transmit_invite function */
-struct sip_invite_param {
-	int addsipheaders;		/*!< Add extra SIP headers */
-	const char *uri_options;	/*!< URI options to add to the URI */
-	const char *vxml_url;		/*!< VXML url for Cisco phones */
-	char *auth;			/*!< Authentication */
-	char *authheader;		/*!< Auth header */
-	enum sip_auth_type auth_type;	/*!< Authentication type */
-	const char *replaces;		/*!< Replaces header for call transfers */
-	int transfer;			/*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */
-};
-
-/*! \brief Structure to save routing information for a SIP session */
-struct sip_route {
-	struct sip_route *next;
-	char hop[0];
-};
-
-/*! \brief Modes for SIP domain handling in the PBX */
-enum domain_mode {
-	SIP_DOMAIN_AUTO,		/*!< This domain is auto-configured */
-	SIP_DOMAIN_CONFIG,		/*!< This domain is from configuration */
-};
-
-/*! \brief Domain data structure. 
-	\note In the future, we will connect this to a configuration tree specific
-	for this domain
-*/
-struct domain {
-	char domain[MAXHOSTNAMELEN];		/*!< SIP domain we are responsible for */
-	char context[AST_MAX_EXTENSION];	/*!< Incoming context for this domain */
-	enum domain_mode mode;			/*!< How did we find this domain? */
-	AST_LIST_ENTRY(domain) list;		/*!< List mechanics */
-};
-
 static AST_LIST_HEAD_STATIC(domain_list, domain);	/*!< The SIP domain list */
-
-
-/*! \brief sip_history: Structure for saving transactions within a SIP dialog */
-struct sip_history {
-	AST_LIST_ENTRY(sip_history) list;
-	char event[0];	/* actually more, depending on needs */
-};
-
 AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
-
-/*! \brief sip_auth: Creadentials for authentication to other SIP services */
-struct sip_auth {
-	char realm[AST_MAX_EXTENSION];  /*!< Realm in which these credentials are valid */
-	char username[256];             /*!< Username */
-	char secret[256];               /*!< Secret */
-	char md5secret[256];            /*!< MD5Secret */
-	struct sip_auth *next;          /*!< Next auth structure in list */
-};
-
-/*--- Various flags for the flags field in the pvt structure */
-#define SIP_ALREADYGONE		(1 << 0)	/*!< Whether or not we've already been destroyed by our peer */
-#define SIP_NEEDDESTROY		(1 << 1)	/*!< if we need to be destroyed by the monitor thread */
-#define SIP_NOVIDEO		(1 << 2)	/*!< Didn't get video in invite, don't offer */
-#define SIP_RINGING		(1 << 3)	/*!< Have sent 180 ringing */
-#define SIP_PROGRESS_SENT	(1 << 4)	/*!< Have sent 183 message progress */
-#define SIP_NEEDREINVITE	(1 << 5)	/*!< Do we need to send another reinvite? */
-#define SIP_PENDINGBYE		(1 << 6)	/*!< Need to send bye after we ack? */
-#define SIP_GOTREFER		(1 << 7)	/*!< Got a refer? */
-#define SIP_PROMISCREDIR	(1 << 8)	/*!< Promiscuous redirection */
-#define SIP_TRUSTRPID		(1 << 9)	/*!< Trust RPID headers? */
-#define SIP_USEREQPHONE		(1 << 10)	/*!< Add user=phone to numeric URI. Default off */
-#define SIP_REALTIME		(1 << 11)	/*!< Flag for realtime users */
-#define SIP_USECLIENTCODE	(1 << 12)	/*!< Trust X-ClientCode info message */
-#define SIP_OUTGOING		(1 << 13)	/*!< Is this an outgoing call? */
-#define SIP_CAN_BYE		(1 << 14)	/*!< Can we send BYE on this dialog? */
-#define SIP_DEFER_BYE_ON_TRANSFER	(1 << 15)	/*!< Do not hangup at first ast_hangup */
-#define SIP_DTMF		(3 << 16)	/*!< DTMF Support: four settings, uses two bits */
-#define SIP_DTMF_RFC2833	(0 << 16)	/*!< DTMF Support: RTP DTMF - "rfc2833" */
-#define SIP_DTMF_INBAND		(1 << 16)	/*!< DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
-#define SIP_DTMF_INFO		(2 << 16)	/*!< DTMF Support: SIP Info messages - "info" */
-#define SIP_DTMF_AUTO		(3 << 16)	/*!< DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
-/* NAT settings */
-#define SIP_NAT			(3 << 18)	/*!< four settings, uses two bits */
-#define SIP_NAT_NEVER		(0 << 18)	/*!< No nat support */
-#define SIP_NAT_RFC3581		(1 << 18)	/*!< NAT RFC3581 */
-#define SIP_NAT_ROUTE		(2 << 18)	/*!< NAT Only ROUTE */
-#define SIP_NAT_ALWAYS		(3 << 18)	/*!< NAT Both ROUTE and RFC3581 */
-/* re-INVITE related settings */
-#define SIP_REINVITE		(7 << 20)	/*!< three bits used */
-#define SIP_CAN_REINVITE	(1 << 20)	/*!< allow peers to be reinvited to send media directly p2p */
-#define SIP_CAN_REINVITE_NAT	(2 << 20)	/*!< allow media reinvite when new peer is behind NAT */
-#define SIP_REINVITE_UPDATE	(4 << 20)	/*!< use UPDATE (RFC3311) when reinviting this peer */
-/* "insecure" settings */
-#define SIP_INSECURE_PORT	(1 << 23)	/*!< don't require matching port for incoming requests */
-#define SIP_INSECURE_INVITE	(1 << 24)	/*!< don't require authentication for incoming INVITEs */
-/* Sending PROGRESS in-band settings */
-#define SIP_PROG_INBAND		(3 << 25)	/*!< three settings, uses two bits */
-#define SIP_PROG_INBAND_NEVER	(0 << 25)
-#define SIP_PROG_INBAND_NO	(1 << 25)
-#define SIP_PROG_INBAND_YES	(2 << 25)
-#define SIP_FREE_BIT		(1 << 27)	/*!< Undefined bit - not in use */
-#define SIP_CALL_LIMIT		(1 << 28)	/*!< Call limit enforced for this call */
-#define SIP_SENDRPID		(1 << 29)	/*!< Remote Party-ID Support */
-#define SIP_INC_COUNT		(1 << 30)	/*!< Did this connection increment the counter of in-use calls? */
-#define SIP_G726_NONSTANDARD	(1 << 31)	/*!< Use non-standard packing for G726-32 data */
-
-#define SIP_FLAGS_TO_COPY \
-	(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
-	 SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
-	 SIP_USEREQPHONE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE)
-
-/*--- a new page of flags (for flags[1] */
-/* realtime flags */
-#define SIP_PAGE2_RTCACHEFRIENDS	(1 << 0)
-#define SIP_PAGE2_RTUPDATE		(1 << 1)
-#define SIP_PAGE2_RTAUTOCLEAR		(1 << 2)
-#define SIP_PAGE2_RT_FROMCONTACT 	(1 << 4)
-#define SIP_PAGE2_RTSAVE_SYSNAME 	(1 << 5)
-/* Space for addition of other realtime flags in the future */
-#define SIP_PAGE2_IGNOREREGEXPIRE	(1 << 10)
-#define SIP_PAGE2_DEBUG			(3 << 11)
-#define SIP_PAGE2_DEBUG_CONFIG 		(1 << 11)
-#define SIP_PAGE2_DEBUG_CONSOLE 	(1 << 12)
-#define SIP_PAGE2_DYNAMIC		(1 << 13)	/*!< Dynamic Peers register with Asterisk */
-#define SIP_PAGE2_SELFDESTRUCT		(1 << 14)	/*!< Automatic peers need to destruct themselves */
-#define SIP_PAGE2_VIDEOSUPPORT		(1 << 15)
-#define SIP_PAGE2_ALLOWSUBSCRIBE	(1 << 16)	/*!< Allow subscriptions from this peer? */
-#define SIP_PAGE2_ALLOWOVERLAP		(1 << 17)	/*!< Allow overlap dialing ? */
-#define SIP_PAGE2_SUBSCRIBEMWIONLY	(1 << 18)	/*!< Only issue MWI notification if subscribed to */
-#define SIP_PAGE2_INC_RINGING		(1 << 19)	/*!< Did this connection increment the counter of in-use calls? */
-#define SIP_PAGE2_T38SUPPORT		(7 << 20)	/*!< T38 Fax Passthrough Support */
-#define SIP_PAGE2_T38SUPPORT_UDPTL	(1 << 20)	/*!< 20: T38 Fax Passthrough Support */
-#define SIP_PAGE2_T38SUPPORT_RTP	(2 << 20)	/*!< 21: T38 Fax Passthrough Support */
-#define SIP_PAGE2_T38SUPPORT_TCP	(4 << 20)	/*!< 22: T38 Fax Passthrough Support */
-#define SIP_PAGE2_CALL_ONHOLD		(3 << 23)	/*!< Call states */
-#define SIP_PAGE2_CALL_ONHOLD_ONEDIR	(1 << 23)	/*!< 23: One directional hold */
-#define SIP_PAGE2_CALL_ONHOLD_INACTIVE	(2 << 24)	/*!< 24: Inactive  */
-#define SIP_PAGE2_RFC2833_COMPENSATE    (1 << 26)
-
-#define SIP_PAGE2_FLAGS_TO_COPY \
-	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE)
-
-/* SIP packet flags */
-#define SIP_PKT_DEBUG		(1 << 0)	/*!< Debug this packet */
-#define SIP_PKT_WITH_TOTAG	(1 << 1)	/*!< This packet has a to-tag */
-#define SIP_PKT_IGNORE 		(1 << 2)	/*!< This is a re-transmit, ignore it */
-#define SIP_PKT_IGNORE_RESP	(1 << 3)	/*!< Resp ignore - ??? */
-#define SIP_PKT_IGNORE_REQ	(1 << 4)	/*!< Req ignore - ??? */
-
-/* T.38 set of flags */
-#define T38FAX_FILL_BIT_REMOVAL		(1 << 0)	/*!< Default: 0 (unset)*/
-#define T38FAX_TRANSCODING_MMR			(1 << 1)	/*!< Default: 0 (unset)*/
-#define T38FAX_TRANSCODING_JBIG		(1 << 2)	/*!< Default: 0 (unset)*/
-/* Rate management */
-#define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF	(0 << 3)
-#define T38FAX_RATE_MANAGEMENT_LOCAL_TCF	(1 << 3)	/*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */
-/* UDP Error correction */
-#define T38FAX_UDP_EC_NONE			(0 << 4)	/*!< two bits, if unset NO t38UDPEC field in T38 SDP*/
-#define T38FAX_UDP_EC_FEC			(1 << 4)	/*!< Set for t38UDPFEC */
-#define T38FAX_UDP_EC_REDUNDANCY		(2 << 4)	/*!< Set for t38UDPRedundancy */
-/* T38 Spec version */
-#define T38FAX_VERSION				(3 << 6)	/*!< two bits, 2 values so far, up to 4 values max */
-#define T38FAX_VERSION_0			(0 << 6)	/*!< Version 0 */
-#define T38FAX_VERSION_1			(1 << 6)	/*!< Version 1 */
-/* Maximum Fax Rate */
-#define T38FAX_RATE_2400			(1 << 8)	/*!< 2400 bps t38FaxRate */
-#define T38FAX_RATE_4800			(1 << 9)	/*!< 4800 bps t38FaxRate */
-#define T38FAX_RATE_7200			(1 << 10)	/*!< 7200 bps t38FaxRate */
-#define T38FAX_RATE_9600			(1 << 11)	/*!< 9600 bps t38FaxRate */
-#define T38FAX_RATE_12000			(1 << 12)	/*!< 12000 bps t38FaxRate */
-#define T38FAX_RATE_14400			(1 << 13)	/*!< 14400 bps t38FaxRate */
 
 /*!< 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;
-
-#define sipdebug		ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)
-#define sipdebug_config		ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)
-#define sipdebug_console	ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)
-
-/*! \brief T38 States for a call */
-enum t38state {
-        T38_DISABLED = 0,                /*!< Not enabled */
-        T38_LOCAL_DIRECT,                /*!< Offered from local */
-        T38_LOCAL_REINVITE,              /*!< Offered from local - REINVITE */
-        T38_PEER_DIRECT,                 /*!< Offered from peer */
-        T38_PEER_REINVITE,               /*!< Offered from peer - REINVITE */
-        T38_ENABLED                      /*!< Negotiated (enabled) */
-};
-
-/*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
-struct t38properties {
-	struct ast_flags t38support;	/*!< Flag for udptl, rtp or tcp support for this session */
-	int capability;			/*!< Our T38 capability */
-	int peercapability;		/*!< Peers T38 capability */
-	int jointcapability;		/*!< Supported T38 capability at both ends */
-	enum t38state state;		/*!< T.38 state */
-};
-
-/*! \brief Parameters to know status of transfer */
-enum referstatus {
-        REFER_IDLE,                    /*!< No REFER is in progress */
-        REFER_SENT,                    /*!< Sent REFER to transferee */
-        REFER_RECEIVED,                /*!< Received REFER from transferer */
-        REFER_CONFIRMED,               /*!< Refer confirmed with a 100 TRYING */
-        REFER_ACCEPTED,                /*!< Accepted by transferee */
-        REFER_RINGING,                 /*!< Target Ringing */
-        REFER_200OK,                   /*!< Answered by transfer target */
-        REFER_FAILED,                  /*!< REFER declined - go on */
-        REFER_NOAUTH                   /*!< We had no auth for REFER */
-};
 
 static const struct c_referstatusstring {
 	enum referstatus status;
@@ -851,256 +405,6 @@
 	{ REFER_NOAUTH,		"Failed - auth failure" }
 } ;
 
-/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed  */
-/* OEJ: Should be moved to string fields */
-struct sip_refer {
-	char refer_to[AST_MAX_EXTENSION];		/*!< Place to store REFER-TO extension */
-	char refer_to_domain[AST_MAX_EXTENSION];	/*!< Place to store REFER-TO domain */
-	char refer_to_urioption[AST_MAX_EXTENSION];	/*!< Place to store REFER-TO uri options */
-	char refer_to_context[AST_MAX_EXTENSION];	/*!< Place to store REFER-TO context */
-	char referred_by[AST_MAX_EXTENSION];		/*!< Place to store REFERRED-BY extension */
-	char referred_by_name[AST_MAX_EXTENSION];	/*!< Place to store REFERRED-BY extension */
-	char refer_contact[AST_MAX_EXTENSION];		/*!< Place to store Contact info from a REFER extension */
-	char replaces_callid[BUFSIZ];			/*!< Replace info: callid */
-	char replaces_callid_totag[BUFSIZ/2];		/*!< Replace info: to-tag */
-	char replaces_callid_fromtag[BUFSIZ/2];		/*!< Replace info: from-tag */
-	struct sip_pvt *refer_call;			/*!< Call we are referring */
-	int attendedtransfer;				/*!< Attended or blind transfer? */
-	int localtransfer;				/*!< Transfer to local domain? */
-	enum referstatus status;			/*!< REFER status */
-};
-
-/*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe  */
-static struct sip_pvt {
-	ast_mutex_t lock;			/*!< Dialog private lock */
-	int method;				/*!< SIP method that opened this dialog */
-	AST_DECLARE_STRING_FIELDS(
-		AST_STRING_FIELD(callid);	/*!< Global CallID */
-		AST_STRING_FIELD(randdata);	/*!< Random data */
-		AST_STRING_FIELD(accountcode);	/*!< Account code */
-		AST_STRING_FIELD(realm);	/*!< Authorization realm */
-		AST_STRING_FIELD(nonce);	/*!< Authorization nonce */
-		AST_STRING_FIELD(opaque);	/*!< Opaque nonsense */
-		AST_STRING_FIELD(qop);		/*!< Quality of Protection, since SIP wasn't complicated enough yet. */
-		AST_STRING_FIELD(domain);	/*!< Authorization domain */
-		AST_STRING_FIELD(from);		/*!< The From: header */
-		AST_STRING_FIELD(useragent);	/*!< User agent in SIP request */
-		AST_STRING_FIELD(exten);	/*!< Extension where to start */
-		AST_STRING_FIELD(context);	/*!< Context for this call */
-		AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */
-		AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */
-		AST_STRING_FIELD(fromdomain);	/*!< Domain to show in the from field */
-		AST_STRING_FIELD(fromuser);	/*!< User to show in the user field */
-		AST_STRING_FIELD(fromname);	/*!< Name to show in the user field */
-		AST_STRING_FIELD(tohost);	/*!< Host we should put in the "to" field */
-		AST_STRING_FIELD(language);	/*!< Default language for this call */
-		AST_STRING_FIELD(mohinterpret);	/*!< MOH class to use when put on hold */
-		AST_STRING_FIELD(mohsuggest);	/*!< MOH class to suggest when putting a peer on hold */
-		AST_STRING_FIELD(rdnis);	/*!< Referring DNIS */
-		AST_STRING_FIELD(theirtag);	/*!< Their tag */
-		AST_STRING_FIELD(username);	/*!< [user] name */
-		AST_STRING_FIELD(peername);	/*!< [peer] name, not set if [user] */
-		AST_STRING_FIELD(authname);	/*!< Who we use for authentication */
-		AST_STRING_FIELD(uri);		/*!< Original requested URI */
-		AST_STRING_FIELD(okcontacturi);	/*!< URI from the 200 OK on INVITE */
-		AST_STRING_FIELD(peersecret);	/*!< Password */
-		AST_STRING_FIELD(peermd5secret);
-		AST_STRING_FIELD(cid_num);	/*!< Caller*ID number */
-		AST_STRING_FIELD(cid_name);	/*!< Caller*ID name */
-		AST_STRING_FIELD(via);		/*!< Via: header */
-		AST_STRING_FIELD(fullcontact);	/*!< The Contact: that the UA registers with us */
-		AST_STRING_FIELD(our_contact);	/*!< Our contact header */
-		AST_STRING_FIELD(rpid);		/*!< Our RPID header */
-		AST_STRING_FIELD(rpid_from);	/*!< Our RPID From header */
-	);
-	unsigned int ocseq;			/*!< Current outgoing seqno */
-	unsigned int icseq;			/*!< Current incoming seqno */
-	ast_group_t callgroup;			/*!< Call group */
-	ast_group_t pickupgroup;		/*!< Pickup group */
-	int lastinvite;				/*!< Last Cseq of invite */
-	struct ast_flags flags[2];		/*!< SIP_ flags */
-	int timer_t1;				/*!< SIP timer T1, ms rtt */
-	unsigned int sipoptions;		/*!< Supported SIP options on the other end */
-	struct ast_codec_pref prefs;		/*!< codec prefs */
-	int capability;				/*!< Special capability (codec) */
-	int jointcapability;			/*!< Supported capability at both ends (codecs ) */
-	int peercapability;			/*!< Supported peer capability */
-	int prefcodec;				/*!< Preferred codec (outbound only) */
-	int noncodeccapability;			/*!< DTMF RFC2833 telephony-event */
-	int redircodecs;			/*!< Redirect codecs */
-	int maxcallbitrate;			/*!< Maximum Call Bitrate for Video Calls */	
-	struct t38properties t38;		/*!< T38 settings */
-	struct sockaddr_in udptlredirip;	/*!< Where our T.38 UDPTL should be going if not to us */
-	struct ast_udptl *udptl;		/*!< T.38 UDPTL session */
-	int callingpres;			/*!< Calling presentation */
-	int authtries;				/*!< Times we've tried to authenticate */
-	int expiry;				/*!< How long we take to expire */
-	long branch;				/*!< The branch identifier of this session */
-	char tag[11];				/*!< Our tag for this session */
-	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 */
-	time_t lastrtprx;			/*!< Last RTP received */
-	time_t lastrtptx;			/*!< Last RTP sent */
-	int rtptimeout;				/*!< RTP timeout time */
-	int rtpholdtimeout;			/*!< RTP timeout when on hold */
-	int rtpkeepalive;			/*!< Send RTP packets for keepalive */
-	struct sockaddr_in recv;		/*!< Received as */
-	struct in_addr ourip;			/*!< Our IP */
-	struct ast_channel *owner;		/*!< Who owns us (if we have an owner) */
-	struct sip_route *route;		/*!< Head of linked list of routing steps (fm Record-Route) */
-	int route_persistant;			/*!< Is this the "real" route? */
-	struct sip_auth *peerauth;		/*!< Realm authentication */
-	int noncecount;				/*!< Nonce-count */
-	char lastmsg[256];			/*!< Last Message sent/received */
-	int amaflags;				/*!< AMA Flags */
-	int pendinginvite;			/*!< Any pending invite ? (seqno of this) */
-	struct sip_request initreq;		/*!< Initial request that opened the SIP dialog */
-	
-	int maxtime;				/*!< Max time for first response */
-	int initid;				/*!< Auto-congest ID if appropriate (scheduler) */
-	int autokillid;				/*!< Auto-kill ID (scheduler) */
-	enum transfermodes allowtransfer;	/*!< REFER: restriction scheme */
-	struct sip_refer *refer;		/*!< REFER: SIP transfer data structure */
-	enum subscriptiontype subscribed;	/*!< SUBSCRIBE: Is this dialog a subscription?  */
-	int stateid;				/*!< SUBSCRIBE: ID for devicestate subscriptions */
-	int laststate;				/*!< SUBSCRIBE: Last known extension state */
-	int dialogver;				/*!< SUBSCRIBE: Version for subscription dialog-info */
-	
-	struct ast_dsp *vad;			/*!< Voice Activation Detection dsp */
-	
-	struct sip_peer *relatedpeer;		/*!< If this dialog is related to a peer, which one 
-							Used in peerpoke, mwi subscriptions */
-	struct sip_registry *registry;		/*!< If this is a REGISTER dialog, 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 */
-	struct sip_history_head *history;	/*!< History of this SIP dialog */
-	struct ast_variable *chanvars;		/*!< Channel variables to set for inbound call */
-	struct sip_pvt *next;			/*!< Next dialog in chain */
-	struct sip_invite_param *options;	/*!< Options for INVITE */
-	int autoframing;
-} *iflist = NULL;
-
-#define FLAG_RESPONSE (1 << 0)
-#define FLAG_FATAL (1 << 1)
-
-/*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission */
-struct sip_pkt {
-	struct sip_pkt *next;			/*!< Next packet in linked list */
-	int retrans;				/*!< Retransmission number */
-	int method;				/*!< SIP method for this packet */
-	int seqno;				/*!< Sequence number */
-	unsigned int flags;			/*!< non-zero if this is a response packet (e.g. 200 OK) */
-	struct sip_pvt *owner;			/*!< Owner AST call */
-	int retransid;				/*!< Retransmission ID */
-	int timer_a;				/*!< SIP timer A, retransmission timer */
-	int timer_t1;				/*!< SIP Timer T1, estimated RTT or 500 ms */
-	int packetlen;				/*!< Length of packet */
-	char data[0];
-};	
-
-/*! \brief Structure for SIP peer data, we place calls to peers if registered  or fixed IP address (host) */
-/* XXX field 'name' must be first otherwise sip_addrcmp() will fail */
-struct sip_peer {
-	ASTOBJ_COMPONENTS(struct sip_peer);	/*!< name, refcount, objflags,  object pointers */
-					/*!< peer->name is the unique name of this object */
-	enum objecttype type;		/*!< SIP_PEER or SIP_USER */
-	char secret[80];		/*!< Password */
-	char md5secret[80];		/*!< Password in MD5 */
-	struct sip_auth *auth;		/*!< Realm authentication list */
-	char context[AST_MAX_CONTEXT];	/*!< Default context for incoming calls */
-	char subscribecontext[AST_MAX_CONTEXT];	/*!< Default context for subscriptions */
-	char username[80];		/*!< Temporary username until registration */ 
-	char accountcode[AST_MAX_ACCOUNT_CODE];	/*!< Account code */
-	int amaflags;			/*!< AMA Flags (for billing) */
-	char tohost[MAXHOSTNAMELEN];	/*!< If not dynamic, IP address */
-	char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */
-	char fromuser[80];		/*!< From: user when calling this peer */
-	char fromdomain[MAXHOSTNAMELEN];	/*!< From: domain when calling this peer */
-	char fullcontact[256];		/*!< Contact registered with us (not in sip.conf) */
-	char cid_num[80];		/*!< Caller ID num */
-	char cid_name[80];		/*!< Caller ID name */
-	int callingpres;		/*!< Calling id presentation */
-	int inUse;			/*!< Number of calls in use */
-	int inRinging;			/*!< Number of calls ringing */
-	int onHold;                     /*!< Peer has someone on hold */
-	int call_limit;			/*!< Limit of concurrent calls */
-	enum transfermodes allowtransfer;	/*! SIP Refer restriction scheme */
-	char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
-	char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
-	char language[MAX_LANGUAGE];	/*!<  Default language for prompts */
-	char mohinterpret[MAX_MUSICCLASS];/*!<  Music on Hold class */
-	char mohsuggest[MAX_MUSICCLASS];/*!<  Music on Hold class */
-	char useragent[256];		/*!<  User agent in SIP request (saved from registration) */
-	struct ast_codec_pref prefs;	/*!<  codec prefs */
-	int lastmsgssent;
-	time_t	lastmsgcheck;		/*!<  Last time we checked for MWI */
-	unsigned int sipoptions;	/*!<  Supported SIP options */
-	struct ast_flags flags[2];	/*!<  SIP_ flags */
-	int expire;			/*!<  When to expire this peer registration */
-	int capability;			/*!<  Codec capability */
-	int rtptimeout;			/*!<  RTP timeout */
-	int rtpholdtimeout;		/*!<  RTP Hold Timeout */
-	int rtpkeepalive;		/*!<  Send RTP packets for keepalive */
-	ast_group_t callgroup;		/*!<  Call group */
-	ast_group_t pickupgroup;	/*!<  Pickup group */
-	struct ast_dnsmgr_entry *dnsmgr;/*!<  DNS refresh manager for peer */
-	struct sockaddr_in addr;	/*!<  IP address of peer */
-	int maxcallbitrate;		/*!< Maximum Bitrate for a video call */
-	
-	/* Qualification */
-	struct sip_pvt *call;		/*!<  Call pointer */
-	int pokeexpire;			/*!<  When to expire poke (qualify= checking) */
-	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;	/*!<  Default IP address, used until registration */
-	struct ast_ha *ha;		/*!<  Access control list */
-	struct ast_variable *chanvars;	/*!<  Variables to set for channel created by user */
-	struct sip_pvt *mwipvt;		/*!<  Subscription for MWI */
-	int lastmsg;
-	int autoframing;
-};
-
-
-
-/*! \brief Registrations with other SIP proxies */
-struct sip_registry {
-	ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1);
-	AST_DECLARE_STRING_FIELDS(
-		AST_STRING_FIELD(callid);	/*!< Global Call-ID */
-		AST_STRING_FIELD(realm);	/*!< Authorization realm */
-		AST_STRING_FIELD(nonce);	/*!< Authorization nonce */
-		AST_STRING_FIELD(opaque);	/*!< Opaque nonsense */
-		AST_STRING_FIELD(qop);		/*!< Quality of Protection, since SIP wasn't complicated enough yet. */
-		AST_STRING_FIELD(domain);	/*!< Authorization domain */
-		AST_STRING_FIELD(username);	/*!< Who we are registering as */
-		AST_STRING_FIELD(authuser);	/*!< Who we *authenticate* as */
-		AST_STRING_FIELD(hostname);	/*!< Domain or host we register to */
-		AST_STRING_FIELD(secret);	/*!< Password in clear text */	
-		AST_STRING_FIELD(md5secret);	/*!< Password in md5 */
-		AST_STRING_FIELD(contact);	/*!< Contact extension */
-		AST_STRING_FIELD(random);
-	);
-	int portno;			/*!<  Optional port override */
-	int expire;			/*!< Sched ID of expiration */
-	int regattempts;		/*!< Number of attempts (since the last success) */
-	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 dialog" in progress */
-	enum sipregistrystate regstate;	/*!< Registration state (see above) */
-	time_t regtime;		/*!< Last succesful registration time */
-	int callid_valid;		/*!< 0 means we haven't chosen callid for this registry yet. */
-	unsigned int ocseq;		/*!< Sequence number we got to for REGISTERs for this registry */
-	struct sockaddr_in us;		/*!< Who the server thinks we are */
-	int noncecount;			/*!< Nonce-count */
-	char lastmsg[256];		/*!< Last Message sent/received */
-};
-
 /* --- Linked lists of various objects --------*/
 
 /*! \brief  The user list: Users and friends */
@@ -1121,7 +425,6 @@
 
 /*! \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 */

Modified: team/oej/codename-pineapple/channels/sip3/sip3.h
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3.h?rev=44743&r1=44742&r2=44743&view=diff
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3.h (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3.h Sat Oct  7 16:36:25 2006
@@ -59,4 +59,708 @@
                                                     below EXPIRY_GUARD_LIMIT */
 #define DEFAULT_EXPIRY 900                          /*!< Expire slowly */
 
+#define DEFAULT_QUALIFY_MAXMS                2000             /*!< Qualification: Must be faster than 2 seconds by default */
+#define DEFAULT_QUALIFY_FREQ_OK      60 * 1000        /*!< Qualification: How often to check for the host to be up */
+			
+#define DEFAULT_QUALIFY_FREQ_NOTOK   10 * 1000        /*!< Qualification: How often to check, if the host is down... */
+
+#define DEFAULT_RETRANS              1000             /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
+#define MAX_RETRANS                  6                /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
+#define SIP_TRANS_TIMEOUT            32000            /*!< SIP request timeout (rfc 3261) 64*T1 
+                                                      \todo Use known T1 for timeout (peerpoke)
+                                                      */
+#define DEFAULT_TRANS_TIMEOUT        -1               /* Use default SIP transaction timeout */
+#define MAX_AUTHTRIES                3                /*!< Try authentication three times, then fail */
+
+#define SIP_MAX_HEADERS              64               /*!< Max amount of SIP headers to read */
+#define SIP_MAX_LINES                64               /*!< Max amount of lines in SIP attachment (like SDP) */
+#define SIP_MAX_PACKET               4096             /*!< Also from RFC 3261 (2543), should sub headers tho */
+
+#define INITIAL_CSEQ                 101              /*!< our initial sip sequence number */
+
+#define RTP 	1

[... 686 lines stripped ...]


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