[asterisk-commits] twilson: branch group/srtp_reboot r244645 - in /team/group/srtp_reboot: ./ ch...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Feb 3 17:46:48 CST 2010
Author: twilson
Date: Wed Feb 3 17:46:43 2010
New Revision: 244645
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=244645
Log:
Fix merge breakage
Added:
team/group/srtp_reboot/channels/sip/
- copied from r244598, trunk/channels/sip/
team/group/srtp_reboot/channels/sip/config_parser.c
- copied unchanged from r244598, trunk/channels/sip/config_parser.c
team/group/srtp_reboot/channels/sip/include/
- copied from r244598, trunk/channels/sip/include/
team/group/srtp_reboot/channels/sip/include/config_parser.h
- copied unchanged from r244598, trunk/channels/sip/include/config_parser.h
team/group/srtp_reboot/channels/sip/include/reqresp_parser.h
- copied unchanged from r244598, trunk/channels/sip/include/reqresp_parser.h
team/group/srtp_reboot/channels/sip/include/sip.h
- copied, changed from r244598, trunk/channels/sip/include/sip.h
team/group/srtp_reboot/channels/sip/include/sip_utils.h
- copied unchanged from r244598, trunk/channels/sip/include/sip_utils.h
team/group/srtp_reboot/channels/sip/reqresp_parser.c
- copied unchanged from r244598, trunk/channels/sip/reqresp_parser.c
Modified:
team/group/srtp_reboot/ (props changed)
team/group/srtp_reboot/CHANGES
team/group/srtp_reboot/channels/Makefile
team/group/srtp_reboot/channels/chan_sip.c
team/group/srtp_reboot/main/features.c
Propchange: team/group/srtp_reboot/
------------------------------------------------------------------------------
automerge = *
Propchange: team/group/srtp_reboot/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed Feb 3 17:46:43 2010
@@ -1,1 +1,1 @@
-/trunk:1-244568
+/trunk:1-244637
Modified: team/group/srtp_reboot/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/group/srtp_reboot/CHANGES?view=diff&rev=244645&r1=244644&r2=244645
==============================================================================
--- team/group/srtp_reboot/CHANGES (original)
+++ team/group/srtp_reboot/CHANGES Wed Feb 3 17:46:43 2010
@@ -1300,6 +1300,8 @@
pickup
* Added parkedcallrecording option to control one-touch record w/ parking
pickup
+ * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
+ parkedcalltransfers option support for multiple parking lots.
* Added BRIDGE_FEATURES variable to set available features for a channel
* The built-in method for doing attended transfers has been updated to
include some new options that allow you to have the transferee sent
Modified: team/group/srtp_reboot/channels/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/group/srtp_reboot/channels/Makefile?view=diff&rev=244645&r1=244644&r2=244645
==============================================================================
--- team/group/srtp_reboot/channels/Makefile (original)
+++ team/group/srtp_reboot/channels/Makefile Wed Feb 3 17:46:43 2010
@@ -69,6 +69,7 @@
rm -f h323/Makefile
$(if $(filter chan_iax2,$(EMBEDDED_MODS)),modules.link,chan_iax2.so): iax2-parser.o iax2-provision.o
+$(if $(filter chan_sip,$(EMBEDDED_MODS)),modules.link,chan_sip.so): sip/config_parser.o sip/reqresp_parser.o
$(if $(filter chan_dahdi,$(EMBEDDED_MODS)),modules.link,chan_dahdi.so): sig_analog.o sig_pri.o
ifneq ($(filter chan_h323,$(EMBEDDED_MODS)),)
Modified: team/group/srtp_reboot/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/group/srtp_reboot/channels/chan_sip.c?view=diff&rev=244645&r1=244644&r2=244645
==============================================================================
--- team/group/srtp_reboot/channels/chan_sip.c (original)
+++ team/group/srtp_reboot/channels/chan_sip.c Wed Feb 3 17:46:43 2010
@@ -218,7 +218,6 @@
#include "asterisk/paths.h" /* need ast_config_AST_SYSTEM_NAME */
#include "asterisk/lock.h"
-#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
@@ -230,7 +229,6 @@
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
-#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
@@ -239,8 +237,6 @@
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
-#include "asterisk/astobj.h"
-#include "asterisk/test.h"
/*
Uncomment the define below, if you are having refcount related memory leaks.
With this uncommented, this module will generate a file, /tmp/refs, which contains
@@ -256,8 +252,6 @@
#include "asterisk/astobj2.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
-#include "asterisk/linkedlists.h"
-#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/netsock.h"
#include "asterisk/localtime.h"
@@ -266,10 +260,12 @@
#include "asterisk/translate.h"
#include "asterisk/ast_version.h"
#include "asterisk/event.h"
-#include "asterisk/tcptls.h"
#include "asterisk/stun.h"
#include "asterisk/cel.h"
-#include "asterisk/strings.h"
+#include "sip/include/sip.h"
+#include "sip/include/config_parser.h"
+#include "sip/include/reqresp_parser.h"
+#include "sip/include/sip_utils.h"
#include "sip_srtp.h"
#include "sdp_crypto.h"
@@ -549,82 +545,10 @@
</manager>
***/
-#ifndef FALSE
-#define FALSE 0
-#endif
-
-#ifndef TRUE
-#define TRUE 1
-#endif
-
-/* Arguments for find_peer */
-#define FINDUSERS (1 << 0)
-#define FINDPEERS (1 << 1)
-#define FINDALLDEVICES (FINDUSERS | FINDPEERS)
-
-#define SIPBUFSIZE 512 /*!< Buffer size for many operations */
-
-#define XMIT_ERROR -2
-
-#define SIP_RESERVED ";/?:@&=+$,# " /*!< Reserved characters in the username part of the URI */
-
-#define DEFAULT_DEFAULT_EXPIRY 120
-#define DEFAULT_MIN_EXPIRY 60
-#define DEFAULT_MAX_EXPIRY 3600
-#define DEFAULT_MWI_EXPIRY 3600
-#define DEFAULT_REGISTRATION_TIMEOUT 20
-#define DEFAULT_MAX_FORWARDS "70"
-
-/* guard limit must be larger than guard secs */
-/* guard min must be < 1000, and should be >= 250 */
-#define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
-#define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
- EXPIRY_GUARD_SECS */
-#define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
- GUARD_PCT turns out to be lower than this, it
- will use this time instead.
- This is in milliseconds. */
-#define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
- below EXPIRY_GUARD_LIMIT */
-#define DEFAULT_EXPIRY 900 /*!< Expire slowly */
-
static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
static int mwi_expiry = DEFAULT_MWI_EXPIRY;
-
-#define DEFAULT_QUALIFY_GAP 100
-#define DEFAULT_QUALIFY_PEERS 1
-
-
-#define CALLERID_UNKNOWN "Anonymous"
-#define FROMDOMAIN_INVALID "anonymous.invalid"
-
-#define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
-#define DEFAULT_QUALIFYFREQ 60 * 1000 /*!< Qualification: How often to check for the host to be up */
-#define DEFAULT_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 DEFAULT_TIMER_T1 500 /*!< SIP timer T1 (according to RFC 3261) */
-#define SIP_TRANS_TIMEOUT 64 * DEFAULT_TIMER_T1 /*!< 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 PROVIS_KEEPALIVE_TIMEOUT 60000 /*!< How long to wait before retransmitting a provisional response (rfc 3261 13.3.1.1) */
-#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_MIN_PACKET 4096 /*!< Initialize size of memory to allocate for packets */
-#define MAX_HISTORY_ENTRIES 50 /*!< Max entires in the history list for a sip_pvt */
-
-#define INITIAL_CSEQ 101 /*!< Our initial sip sequence number */
-
-#define DEFAULT_MAX_SE 1800 /*!< Session-Timer Default Session-Expires period (RFC 4028) */
-#define DEFAULT_MIN_SE 90 /*!< Session-Timer Default Min-SE period (RFC 4028) */
-
-#define SDP_MAX_RTPMAP_CODECS 32 /*!< Maximum number of codecs allowed in received SDP */
/*! \brief Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
@@ -634,47 +558,13 @@
.resync_threshold = -1,
.impl = ""
};
-static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration */
-
-static const char config[] = "sip.conf"; /*!< Main configuration file */
-static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
-
-#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 */
-};
-
-
-/*! \brief The result of a lot of functions */
-enum sip_result {
- AST_SUCCESS = 0, /*!< FALSE means success, funny enough */
- AST_FAILURE = -1, /*!< Failure code */
-};
-
-/*! \brief States for the INVITE transaction, not the dialog
- \note this is for the INVITE that sets up the dialog
-*/
-enum invitestates {
- INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
- INV_CALLING = 1, /*!< Invite sent, no answer */
- INV_PROCEEDING = 2, /*!< We got/sent 1xx message */
- INV_EARLY_MEDIA = 3, /*!< We got 18x message with to-tag back */
- INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
- INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */
- INV_TERMINATED = 6, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
- The only way out of this is a BYE from one side */
- INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
-};
+static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration */
+
+static const char config[] = "sip.conf"; /*!< Main configuration file */
+static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
/*! \brief Readable descriptions of device states.
- \note Should be aligned to above table as index */
+ * \note Should be aligned to above table as index */
static const struct invstate2stringtable {
const enum invitestates state;
const char *desc;
@@ -689,52 +579,11 @@
{INV_CANCELLED, "Cancelled"}
};
-/*! \brief When sending a SIP message, we can send with a few options, depending on
- type of SIP request. UNRELIABLE is moslty used for responses to repeated requests,
- where the original response would be sent RELIABLE in an INVITE transaction */
-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 */
-};
-
-/*! \brief Results from the parse_register() function */
-enum parse_register_result {
- PARSE_REGISTER_DENIED,
- PARSE_REGISTER_FAILED,
- PARSE_REGISTER_UPDATE,
- PARSE_REGISTER_QUERY,
-};
-
-/*! \brief Type of subscription, based on the packages we do support, see \ref subscription_types */
-enum subscriptiontype {
- NONE = 0,
- XPIDF_XML,
- DIALOG_INFO_XML,
- CPIM_PIDF_XML,
- PIDF_XML,
- MWI_NOTIFICATION
-};
-
-/*! \brief The number of media types in enum \ref media_type below. */
-#define OFFERED_MEDIA_COUNT 4
-
-/*! \brief Media types generate different "dummy answers" for not accepting the offer of
- a media stream. We need to add definitions for each RTP profile. Secure RTP is not
- the same as normal RTP and will require a new definition */
-enum media_type {
- SDP_AUDIO, /*!< RTP/AVP Audio */
- SDP_VIDEO, /*!< RTP/AVP Video */
- SDP_IMAGE, /*!< Image udptl, not TCP or RTP */
- SDP_TEXT, /*!< RTP/AVP Realtime Text */
-};
-
/*! \brief Subscription types that we support. We support
- - dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
- - SIMPLE presence used for device status
- - Voicemail notification subscriptions
-*/
+ * - dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
+ * - SIMPLE presence used for device status
+ * - Voicemail notification subscriptions
+ */
static const struct cfsubscription_types {
enum subscriptiontype type;
const char * const event;
@@ -750,237 +599,42 @@
{ MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
};
-
-/*! \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 = 407,
- WWW_AUTH = 401,
-};
-
-/*! \brief Authentication result from check_auth* functions */
-enum check_auth_result {
- AUTH_DONT_KNOW = -100, /*!< no result, need to check further */
- /* XXX maybe this is the same as AUTH_NOT_FOUND */
-
- AUTH_SUCCESSFUL = 0,
- AUTH_CHALLENGE_SENT = 1,
- AUTH_SECRET_FAILED = -1,
- AUTH_USERNAME_MISMATCH = -2,
- AUTH_NOT_FOUND = -3, /*!< returned by register_verify */
- AUTH_FAKE_AUTH = -4,
- AUTH_UNKNOWN_DOMAIN = -5,
- AUTH_PEER_NOT_DYNAMIC = -6,
- AUTH_ACL_FAILED = -7,
- AUTH_BAD_TRANSPORT = -8,
- AUTH_RTP_FAILED = 9,
-};
-
-/*! \brief States for outbound registrations (with register= lines in sip.conf */
-enum sipregistrystate {
- REG_STATE_UNREGISTERED = 0, /*!< We are not registered
- * \note Initial state. We should have a timeout scheduled for the initial
- * (or next) registration transmission, calling sip_reregister
- */
-
- REG_STATE_REGSENT, /*!< Registration request sent
- * \note sent initial request, waiting for an ack or a timeout to
- * retransmit the initial request.
- */
-
- REG_STATE_AUTHSENT, /*!< We have tried to authenticate
- * \note entered after transmit_register with auth info,
- * waiting for an ack.
- */
-
- REG_STATE_REGISTERED, /*!< Registered and done */
-
- REG_STATE_REJECTED, /*!< Registration rejected
- * \note only used when the remote party has an expire larger than
- * our max-expire. This is a final state from which we do not
- * recover (not sure how correctly).
- */
-
- REG_STATE_TIMEOUT, /*!< Registration timed out
- * \note XXX unused */
-
- REG_STATE_NOAUTH, /*!< We have no accepted credentials
- * \note fatal - no chance to proceed */
-
- REG_STATE_FAILED, /*!< Registration failed after several tries
- * \note fatal - no chance to proceed */
-};
-
-/*! \brief Modes in which Asterisk can be configured to run SIP Session-Timers */
-enum st_mode {
- SESSION_TIMER_MODE_INVALID = 0, /*!< Invalid value */
- SESSION_TIMER_MODE_ACCEPT, /*!< Honor inbound Session-Timer requests */
- SESSION_TIMER_MODE_ORIGINATE, /*!< Originate outbound and honor inbound requests */
- SESSION_TIMER_MODE_REFUSE /*!< Ignore inbound Session-Timers requests */
-};
-
-/*! \brief The entity playing the refresher role for Session-Timers */
-enum st_refresher {
- SESSION_TIMER_REFRESHER_AUTO, /*!< Negotiated */
- SESSION_TIMER_REFRESHER_UAC, /*!< Session is refreshed by the UAC */
- SESSION_TIMER_REFRESHER_UAS /*!< Session is refreshed by the UAS */
-};
-
-/*! \brief Define some implemented SIP transports
- \note Asterisk does not support SCTP or UDP/DTLS
-*/
-enum sip_transport {
- SIP_TRANSPORT_UDP = 1, /*!< Unreliable transport for SIP, needs retransmissions */
- SIP_TRANSPORT_TCP = 1 << 1, /*!< Reliable, but unsecure */
- SIP_TRANSPORT_TLS = 1 << 2, /*!< TCP/TLS - reliable and secure transport for signalling */
-};
-
-/*! \brief definition of a sip proxy server
+/*! \brief The core structure to setup dialogs. We parse incoming messages by using
+ * structure and then route the messages according to the type.
*
- * For outbound proxies, a sip_peer will contain a reference to a
- * dynamically allocated instance of a sip_proxy. A sip_pvt may also
- * contain a reference to a peer's outboundproxy, or it may contain
- * a reference to the sip_cfg.outboundproxy.
+ * \note Note that sip_methods[i].id == i must hold or the code breaks
*/
-struct sip_proxy {
- char name[MAXHOSTNAMELEN]; /*!< DNS name of domain/host or IP */
- struct sockaddr_in ip; /*!< Currently used IP address and port */
- time_t last_dnsupdate; /*!< When this was resolved */
- enum sip_transport transport;
- int force; /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
- /* Room for a SRV record chain based on the name */
-};
-
-/*! \brief argument for the 'show channels|subscriptions' callback. */
-struct __show_chan_arg {
- int fd;
- int subscriptions;
- int numchans; /* return value */
-};
-
-
-/*! \brief States whether a SIP message can create a dialog in Asterisk. */
-enum can_create_dialog {
- CAN_NOT_CREATE_DIALOG,
- CAN_CREATE_DIALOG,
- CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
-};
-
-/*! \brief SIP Request methods known by Asterisk
-
- \note 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 sipmethod {
- SIP_UNKNOWN, /*!< Unknown response */
- SIP_RESPONSE, /*!< Not request, response to outbound request */
- SIP_REGISTER, /*!< Registration to the mothership, tell us where you are located */
- SIP_OPTIONS, /*!< Check capabilities of a device, used for "ping" too */
- SIP_NOTIFY, /*!< Status update, Part of the event package standard, result of a SUBSCRIBE or a REFER */
- SIP_INVITE, /*!< Set up a session */
- SIP_ACK, /*!< End of a three-way handshake started with INVITE. */
- SIP_PRACK, /*!< Reliable pre-call signalling. Not supported in Asterisk. */
- SIP_BYE, /*!< End of a session */
- SIP_REFER, /*!< Refer to another URI (transfer) */
- SIP_SUBSCRIBE, /*!< Subscribe for updates (voicemail, session status, device status, presence) */
- SIP_MESSAGE, /*!< Text messaging */
- SIP_UPDATE, /*!< Update a dialog. We can send UPDATE; but not accept it */
- SIP_INFO, /*!< Information updates during a session */
- SIP_CANCEL, /*!< Cancel an INVITE */
- SIP_PUBLISH, /*!< Not supported in Asterisk */
- SIP_PING, /*!< Not supported at all, no standard but still implemented out there */
-};
-
-/*! \brief Settings for the 'notifycid' option, see sip.conf.sample for details. */
-enum notifycid_setting {
- DISABLED = 0,
- ENABLED = 1,
- IGNORE_CONTEXT = 2,
-};
-
-/*! \brief The core structure to setup dialogs. We parse incoming messages by using
- structure and then route the messages according to the type.
-
- \note Note that sip_methods[i].id == i must hold or the code breaks */
static const struct cfsip_methods {
enum sipmethod id;
int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
char * const text;
enum can_create_dialog can_create;
} sip_methods[] = {
- { SIP_UNKNOWN, RTP, "-UNKNOWN-", CAN_CREATE_DIALOG },
- { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
- { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
- { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
- { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
- { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
- { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
- { SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG },
- { SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG },
- { SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG },
- { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE", CAN_CREATE_DIALOG },
- { SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG },
- { SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG },
- { SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG },
- { SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG },
- { SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD },
- { SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
+ { SIP_UNKNOWN, RTP, "-UNKNOWN-",CAN_CREATE_DIALOG },
+ { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
+ { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
+ { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
+ { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
+ { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
+ { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
+ { SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG },
+ { SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG },
+ { SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG },
+ { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE",CAN_CREATE_DIALOG },
+ { SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG },
+ { SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG },
+ { SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG },
+ { SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG },
+ { SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD },
+ { SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
};
-
-static unsigned int chan_idx;
-
-/*! 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
-
-/* SIP options */
-#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)
-#define SIP_OPT_FROMCHANGE (1 << 17)
-#define SIP_OPT_RECLISTINV (1 << 18)
-#define SIP_OPT_RECLISTSUB (1 << 19)
-#define SIP_OPT_OUTBOUND (1 << 20)
-#define SIP_OPT_UNKNOWN (1 << 21)
-
/*! \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 {
- int id; /*!< Bitmap ID */
- int supported; /*!< Supported by Asterisk ? */
- char * const text; /*!< Text id, as in standard */
+ int id; /*!< Bitmap ID */
+ int supported; /*!< Supported by Asterisk ? */
+ char * const text; /*!< Text id, as in standard */
} sip_options[] = { /* XXX used in 3 places */
/* RFC3262: PRACK 100% reliability */
{ SIP_OPT_100REL, NOT_SUPPORTED, "100rel" },
@@ -1053,175 +707,29 @@
{ AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"}
};
-static enum AST_REDIRECTING_REASON sip_reason_str_to_code(const char *text)
-{
- enum AST_REDIRECTING_REASON ast = AST_REDIRECTING_REASON_UNKNOWN;
- int i;
-
- for (i = 0; i < ARRAY_LEN(sip_reason_table); ++i) {
- if (!strcasecmp(text, sip_reason_table[i].text)) {
- ast = sip_reason_table[i].code;
- break;
- }
- }
-
- return ast;
-}
-
-static const char *sip_reason_code_to_str(enum AST_REDIRECTING_REASON code)
-{
- if (code >= 0 && code < ARRAY_LEN(sip_reason_table)) {
- return sip_reason_table[code].text;
- }
-
- return "unknown";
-}
-
-/*! \brief SIP Methods we support
- \todo This string should be set dynamically. We only support REFER and SUBSCRIBE if we have
- allowsubscribe and allowrefer on in sip.conf.
-*/
-#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO"
-
-/*! \brief SIP Extensions we support
- \note This should be generated based on the previous array
- in combination with settings.
- \todo We should not have "timer" if it's disabled in the configuration file.
-*/
-#define SUPPORTED_EXTENSIONS "replaces, timer"
-
-/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
-#define STANDARD_SIP_PORT 5060
-/*! \brief Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS */
-#define STANDARD_TLS_PORT 5061
-
-/*! \note in many SIP headers, absence of a port number implies port 5060,
- * and this is why we cannot change the above constant.
- * There is a limited number of places in asterisk where we could,
- * in principle, use a different "default" port number, but
- * we do not support this feature at the moment.
- * You can run Asterisk with SIP on a different port with a configuration
- * option. If you change this value in the source code, the signalling will be incorrect.
- *
- */
-
-/*! \name DefaultValues 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_CONTEXT "default" /*!< The default context for [general] section as well as devices */
-#define DEFAULT_MOHINTERPRET "default" /*!< The default music class */
-#define DEFAULT_MOHSUGGEST ""
-#define DEFAULT_VMEXTEN "asterisk" /*!< Default voicemail extension */
-#define DEFAULT_CALLERID "asterisk" /*!< Default caller ID */
-#define DEFAULT_MWI_FROM ""
-#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
-#define DEFAULT_ALLOWGUEST TRUE
-#define DEFAULT_RTPKEEPALIVE 0 /*!< Default RTPkeepalive setting */
-#define DEFAULT_CALLCOUNTER FALSE /*!< Do not enable call counters by default */
-#define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */
-#define DEFAULT_COMPACTHEADERS FALSE /*!< Send compact (one-character) SIP headers. Default off */
-#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_TEXT 0 /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_COS_SIP 4 /*!< Level 2 class of service for SIP signalling */
-#define DEFAULT_COS_AUDIO 5 /*!< Level 2 class of service for audio media */
-#define DEFAULT_COS_VIDEO 6 /*!< Level 2 class of service for video media */
-#define DEFAULT_COS_TEXT 5 /*!< Level 2 class of service for text media (T.140) */
-#define DEFAULT_ALLOW_EXT_DOM TRUE /*!< Allow external domains */
-#define DEFAULT_REALM "asterisk" /*!< Realm for HTTP digest authentication */
-#define DEFAULT_DOMAINSASREALM FALSE /*!< Use the domain option to guess the realm for registration and invite requests */
-#define DEFAULT_NOTIFYRINGING TRUE /*!< Notify devicestate system on ringing state */
-#define DEFAULT_NOTIFYCID DISABLED /*!< Include CID with ringing notifications */
-#define DEFAULT_PEDANTIC FALSE /*!< Avoid following SIP standards for dialog matching */
-#define DEFAULT_AUTOCREATEPEER FALSE /*!< Don't create peers automagically */
-#define DEFAULT_MATCHEXTERNIPLOCALLY FALSE /*!< Match extern IP locally default setting */
-#define DEFAULT_QUALIFY FALSE /*!< Don't monitor devices */
-#define DEFAULT_CALLEVENTS FALSE /*!< Extra manager SIP call events */
-#define DEFAULT_ALWAYSAUTHREJECT FALSE /*!< Don't reject authentication requests always */
-#define DEFAULT_REGEXTENONQUALIFY 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 */
-#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
-#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
-#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
-#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
-#endif
-/*@}*/
/*! \name DefaultSettings
Default setttings are used as a channel setting and as a default when
configuring devices
*/
/*@{*/
-static char default_language[MAX_LANGUAGE]; /*!< Default language setting for new channels */
-static char default_callerid[AST_MAX_EXTENSION]; /*!< Default caller ID for sip messages */
-static char default_mwi_from[80]; /*!< Default caller ID for MWI updates */
-static char default_fromdomain[AST_MAX_EXTENSION]; /*!< Default domain on outound messages */
-static char default_notifymime[AST_MAX_EXTENSION]; /*!< Default MIME media type for MWI notify messages */
-static char default_vmexten[AST_MAX_EXTENSION]; /*!< Default From Username on MWI updates */
-static int default_qualify; /*!< Default Qualify= setting */
+static char default_language[MAX_LANGUAGE]; /*!< Default language setting for new channels */
+static char default_callerid[AST_MAX_EXTENSION]; /*!< Default caller ID for sip messages */
+static char default_mwi_from[80]; /*!< Default caller ID for MWI updates */
+static char default_fromdomain[AST_MAX_EXTENSION]; /*!< Default domain on outound messages */
+static char default_notifymime[AST_MAX_EXTENSION]; /*!< Default MIME media type for MWI notify messages */
+static char default_vmexten[AST_MAX_EXTENSION]; /*!< Default From Username on MWI updates */
+static int default_qualify; /*!< Default Qualify= setting */
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
+static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
* a bridged channel on hold */
-static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
-static char default_engine[256]; /*!< Default RTP engine */
-static int default_maxcallbitrate; /*!< Maximum bitrate for call */
-static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
-static unsigned int default_transports; /*!< Default Transports (enum sip_transport) that are acceptable */
-static unsigned int default_primary_transport; /*!< Default primary Transport (enum sip_transport) for outbound connections to devices */
-
+static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
+static char default_engine[256]; /*!< Default RTP engine */
+static int default_maxcallbitrate; /*!< Maximum bitrate for call */
+static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
+static unsigned int default_transports; /*!< Default Transports (enum sip_transport) that are acceptable */
+static unsigned int default_primary_transport; /*!< Default primary Transport (enum sip_transport) for outbound connections to devices */
/*@}*/
-
-/*! \name GlobalSettings
- Global settings apply to the channel (often settings you can change in the general section
- of sip.conf
-*/
-/*@{*/
-/*! \brief a place to store all global settings for the sip channel driver
-
- These are settings that will be possibly to apply on a group level later on.
- \note Do not add settings that only apply to the channel itself and can't
- be applied to devices (trunks, services, phones)
-*/
-struct sip_settings {
- int peer_rtupdate; /*!< G: Update database with registration data for peer? */
- int rtsave_sysname; /*!< G: Save system name at registration? */
- int ignore_regexpire; /*!< G: Ignore expiration of peer */
- int rtautoclear; /*!< Realtime ?? */
- int directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */
- int pedanticsipchecking; /*!< Extra checking ? Default off */
- int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
- int srvlookup; /*!< SRV Lookup on or off. Default is on */
- int allowguest; /*!< allow unauthenticated peers to connect? */
- int alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
- int compactheaders; /*!< send compact sip headers */
- int allow_external_domains; /*!< Accept calls to external SIP domains? */
- int callevents; /*!< Whether we send manager events or not */
- int regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
- int matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
- char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
- unsigned int disallowed_methods; /*!< methods that we should never try to use */
- int notifyringing; /*!< Send notifications on ringing */
- int notifyhold; /*!< Send notifications on hold */
- enum notifycid_setting notifycid; /*!< Send CID with ringing notifications */
- enum transfermodes allowtransfer; /*!< SIP Refer restriction scheme */
- int allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
- the global setting is in globals_flags[1] */
- char realm[MAXHOSTNAMELEN]; /*!< Default realm */
- int domainsasrealm; /*!< Use domains lists as realms */
- struct sip_proxy outboundproxy; /*!< Outbound proxy */
- char default_context[AST_MAX_CONTEXT];
- char default_subscribecontext[AST_MAX_CONTEXT];
- struct ast_ha *contact_ha; /*! \brief Global list of addresses dynamic peers are not allowed to use */
- format_t capability; /*!< Supported codecs */
-};
static struct sip_settings sip_cfg; /*!< SIP configuration data.
\note in the future we could have multiple of these (per domain, per device group etc) */
@@ -1232,65 +740,63 @@
ast_uri_decode(str); \
} \
-static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
-
-static int global_relaxdtmf; /*!< Relax DTMF */
-static int global_prematuremediafilter; /*!< Enable/disable premature frames in a call (causing 183 early media) */
-static int global_rtptimeout; /*!< Time out call if no RTP */
-static int global_rtpholdtimeout; /*!< Time out call if no RTP during hold */
-static int global_rtpkeepalive; /*!< Send RTP keepalives */
-static int global_reg_timeout; /*!< Global time between attempts for outbound registrations */
-static int global_regattempts_max; /*!< Registration attempts before giving up */
-static int global_shrinkcallerid; /*!< enable or disable shrinking of caller id */
-static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer
- call-limit to INT_MAX. When we remove the call-limit from the code, we can make it
- with just a boolean flag in the device structure */
-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_text; /*!< IP type of service for text RTP packets */
-static unsigned int global_cos_sip; /*!< 802.1p class of service for SIP packets */
-static unsigned int global_cos_audio; /*!< 802.1p class of service for audio RTP packets */
-static unsigned int global_cos_video; /*!< 802.1p class of service for video RTP packets */
-static unsigned int global_cos_text; /*!< 802.1p class of service for text RTP packets */
-static unsigned int recordhistory; /*!< Record SIP history. Off by default */
-static unsigned int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
-static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
-static char global_sdpsession[AST_MAX_EXTENSION]; /*!< SDP session name for the SIP channel */
-static char global_sdpowner[AST_MAX_EXTENSION]; /*!< SDP owner name for the SIP channel */
-static int global_authfailureevents; /*!< Whether we send authentication failure manager events or not. Default no. */
-static int global_t1; /*!< T1 time */
-static int global_t1min; /*!< T1 roundtrip time minimum */
-static int global_timer_b; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
-static unsigned int global_autoframing; /*!< Turn autoframing on or off. */
-static int global_qualifyfreq; /*!< Qualify frequency */
-static int global_qualify_gap; /*!< Time between our group of peer pokes */
-static int global_qualify_peers; /*!< Number of peers to poke at a given time */
-
+static unsigned int chan_idx; /*!< used in naming sip channel */
+static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
+
[... 2103 lines stripped ...]
More information about the asterisk-commits
mailing list