[svn-commits] oej: branch oej/codename-pineapple r44743 - in
/team/oej/codename-pineapple/c...
svn-commits at lists.digium.com
svn-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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