chan_sip.c File Reference

Implementation of Session Initiation Protocol. More...

#include "asterisk.h"
#include <ctype.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/signal.h>
#include <regex.h>
#include <time.h>
#include "asterisk/network.h"
#include "asterisk/paths.h"
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#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"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj.h"
#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"
#include "asterisk/abstract_jb.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"
#include "asterisk/ast_version.h"
#include "asterisk/event.h"
#include "asterisk/tcptls.h"
#include "asterisk/strings.h"

Include dependency graph for chan_sip.c:

Go to the source code of this file.

Data Structures
struct	__show_chan_arg
	argument for the 'show channels|subscriptions' callback. More...
struct	_map_x_s
	generic struct to map between strings and integers. Fill it with x-s pairs, terminate with an entry with s = NULL; Then you can call map_x_s(...) to map an integer to a string, and map_s_x() for the string -> integer mapping. More...
struct	ast_register_list
	The register list: Other SIP proxies we register with and place calls to. More...
struct	ast_subscription_mwi_list
	The MWI subscription list. More...
struct	cfsip_methods
	The core structure to setup dialogs. We parse incoming messages by using structure and then route the messages according to the type. More...
struct	cfsip_options
	List of well-known SIP options. If we get this in a require, we should check the list and answer accordingly. More...
struct	cfsubscription_types
	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. More...
struct	domain
	Domain data structure. More...
struct	domain_list
struct	find_call_cb_arg
	argument to the helper function to identify a call More...
struct	invstate2stringtable
	Readable descriptions of device states. More...
struct	offered_media
struct	sip_auth
	sip_auth: Credentials for authentication to other SIP services More...
struct	sip_dual
	structure used in transfers More...
struct	sip_history
	sip_history: Structure for saving transactions within a SIP dialog More...
struct	sip_history_head
struct	sip_invite_param
	Parameters to the transmit_invite function. More...
struct	sip_mailbox
	A peer's mailbox. More...
struct	sip_peer
	Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host). More...
struct	sip_pkt
	sip packet - raw format for outbound packets that are sent or scheduled for transmission Packets are linked in a list, whose head is in the struct sip_pvt they belong to. Each packet holds a reference to the parent struct sip_pvt. This structure is allocated in __sip_reliable_xmit() and only for packets that require retransmissions. More...
struct	sip_proxy
	definition of a sip proxy server More...
struct	sip_pvt
	Structure used for each SIP dialog, ie. a call, a registration, a subscribe. Created and initialized by sip_alloc(), the descriptor goes into the list of descriptors (dialoglist). More...
struct	sip_pvt::request_queue
struct	sip_refer
	Structure to handle SIP transfers. Dynamically allocated when needed. More...
struct	sip_registry
	Registrations with other SIP proxies. More...
struct	sip_request
	sip_request: The data grabbed from the UDP socket More...
struct	sip_route
	Structure to save routing information for a SIP session. More...
struct	sip_settings
	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. More...
struct	sip_socket
	The SIP socket definition. More...
struct	sip_st_cfg
	Structure that encapsulates all attributes related to configuration of SIP Session-Timers feature on a per user/peer basis. More...
struct	sip_st_dlg
	Structure that encapsulates all attributes related to running SIP Session-Timers feature on a per dialog basis. More...
struct	sip_subscription_mwi
	Definition of an MWI subscription to another server. More...
struct	sip_threadinfo
	Definition of a thread that handles a socket. More...
struct	sip_via
	Structure to store Via information. More...
struct	t38properties
	T.38 channel settings (at some point we need to make this alloc'ed. More...
struct	tcptls_packet
Object counters @{
Bug: These counters are not handled in a thread-safe way ast_atomic_fetchadd_int() should be used to modify these values.
#define	DEC_CALL_LIMIT   0
#define	DEC_CALL_RINGING   2
#define	INC_CALL_LIMIT   1
#define	INC_CALL_RINGING   3
#define	REQ_OFFSET_TO_STR(req, offset)   (ast_str_buffer((req)->data) + ((req)->offset))
enum	domain_mode { SIP_DOMAIN_AUTO, SIP_DOMAIN_CONFIG }
	Modes for SIP domain handling in the PBX. More...
static int	apeerobjs = 0
static struct ast_flags	global_flags [2] = {{0}}
static int	global_t38_maxdatagram
static struct io_context *	io
static pthread_t	monitor_thread = AST_PTHREADT_NULL
	This is the thread for the monitor which checks for input on the channels which are not currently in use.
static ast_mutex_t	monlock = ((ast_mutex_t) PTHREAD_MUTEX_INITIALIZER )
	Protect the monitoring thread, so only one process can kill or start it, and not when it's doing something critical.
static ast_mutex_t	netlock = ((ast_mutex_t) PTHREAD_MUTEX_INITIALIZER )
static int	regobjs = 0
static int	rpeerobjs = 0
static struct sched_context *	sched
static ast_mutex_t	sip_reload_lock = ((ast_mutex_t) PTHREAD_MUTEX_INITIALIZER )
static int	sip_reloading = FALSE
static enum channelreloadreason	sip_reloadreason
static int *	sipsock_read_id
static int	speerobjs = 0
static char	used_context [AST_MAX_CONTEXT]
Defines
#define	ALLOWED_METHODS   "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO"
	SIP Methods we support.
#define	append_history(p, event, fmt, args...)   append_history_full(p, "%-15s " fmt, event, ## args)
	Append to SIP dialog history.
#define	CALLERID_UNKNOWN   "Anonymous"
#define	CHECK_AUTH_BUF_INITLEN   256
#define	check_request_transport(peer, tmpl)
	generic function for determining if a correct transport is being used to contact a peer
#define	DEFAULT_AUTHLIMIT   100
#define	DEFAULT_AUTHTIMEOUT   30
#define	DEFAULT_DEFAULT_EXPIRY   120
#define	DEFAULT_EXPIRY   900
#define	DEFAULT_FREQ_NOTOK   10 * 1000
#define	DEFAULT_MAX_EXPIRY   3600
#define	DEFAULT_MAX_FORWARDS   "70"
#define	DEFAULT_MAX_SE   1800
#define	DEFAULT_MAXMS   2000
#define	DEFAULT_MIN_EXPIRY   60
#define	DEFAULT_MIN_SE   90
#define	DEFAULT_MWI_EXPIRY   3600
#define	DEFAULT_QUALIFY_GAP   100
#define	DEFAULT_QUALIFY_PEERS   1
#define	DEFAULT_QUALIFYFREQ   60 * 1000
#define	DEFAULT_REGISTRATION_TIMEOUT   20
#define	DEFAULT_RETRANS   1000
#define	DEFAULT_TIMER_T1   500
#define	DEFAULT_TRANS_TIMEOUT   -1
#define	EXPIRY_GUARD_LIMIT   30
#define	EXPIRY_GUARD_MIN   500
#define	EXPIRY_GUARD_PCT   0.20
#define	EXPIRY_GUARD_SECS   15
#define	FALSE   0
#define	FINDALLDEVICES   (FINDUSERS | FINDPEERS)
#define	FINDPEERS   (1 << 1)
#define	FINDUSERS   (1 << 0)
#define	FORMAT   "%-15.15s %-15.15s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s\n"
#define	FORMAT   "%-30.30s %-12.12s %-10.10s %-10.10s\n"
#define	FORMAT   "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.6u %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.6u\n"
#define	FORMAT   "%-30.30s %-6.6s %-12.12s %8d %-20.20s %-25.25s\n"
#define	FORMAT   "%-40.40s %-20.20s %-16.16s\n"
#define	FORMAT   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %s\n"
#define	FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"
#define	FORMAT   "%-30.30s %-6d %-9.9s %-6.6s\n"
#define	FORMAT   "%-25.25s %-15.15s %-15.15s \n"
#define	FORMAT2   "%-15.15s %-15.15s %-15.15s %-15.15s %-7.7s %-15.15s %-6.6s\n"
#define	FORMAT2   "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s ( %%) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
#define	FORMAT2   "%-30.30s %-6.6s %-12.12s %8.8s %-20.20s %-25.25s\n"
#define	FORMAT2   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %s\n"
#define	FORMAT2   "%-30.30s %3.6s %9.9s %6.6s\n"
#define	FORMAT2   "%-25.25s %-15.15s %-15.15s \n"
#define	FORMAT3   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"
#define	FORMAT4   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"
#define	FROMDOMAIN_INVALID   "anonymous.invalid"
#define	INITIAL_CSEQ   101
#define	IS_SIP_TECH(t)   ((t) == &sip_tech || (t) == &sip_tech_info)
#define	MAX_AUTHTRIES   3
#define	MAX_HISTORY_ENTRIES   50
#define	MAX_RETRANS   6
#define	NO_RTP   0
#define	NOT_SUPPORTED   0
#define	PROVIS_KEEPALIVE_TIMEOUT   60000
#define	RTP   1
#define	SDP_MAX_RTPMAP_CODECS   32
#define	SIP_MAX_HEADERS   64
#define	SIP_MAX_LINES   256
#define	SIP_MIN_PACKET   4096
#define	SIP_OPT_100REL   (1 << 1)
#define	SIP_OPT_EARLY_SESSION   (1 << 3)
#define	SIP_OPT_EVENTLIST   (1 << 11)
#define	SIP_OPT_FROMCHANGE   (1 << 17)
#define	SIP_OPT_GRUU   (1 << 12)
#define	SIP_OPT_HISTINFO   (1 << 15)
#define	SIP_OPT_JOIN   (1 << 4)
#define	SIP_OPT_NOREFERSUB   (1 << 14)
#define	SIP_OPT_OUTBOUND   (1 << 20)
#define	SIP_OPT_PATH   (1 << 5)
#define	SIP_OPT_PRECONDITION   (1 << 7)
#define	SIP_OPT_PREF   (1 << 6)
#define	SIP_OPT_PRIVACY   (1 << 8)
#define	SIP_OPT_RECLISTINV   (1 << 18)
#define	SIP_OPT_RECLISTSUB   (1 << 19)
#define	SIP_OPT_REPLACES   (1 << 0)
#define	SIP_OPT_RESPRIORITY   (1 << 16)
#define	SIP_OPT_SDP_ANAT   (1 << 9)
#define	SIP_OPT_SEC_AGREE   (1 << 10)
#define	SIP_OPT_TARGET_DIALOG   (1 << 13)
#define	SIP_OPT_TIMER   (1 << 2)
#define	SIP_OPT_UNKNOWN   (1 << 21)
#define	sip_pvt_lock(x)   ao2_lock(x)
#define	sip_pvt_trylock(x)   ao2_trylock(x)
#define	sip_pvt_unlock(x)   ao2_unlock(x)
#define	SIP_RESERVED   ";/?:@&=+$,# "
#define	SIP_TRANS_TIMEOUT   64 * DEFAULT_TIMER_T1
#define	SIPBUFSIZE   512
#define	STANDARD_SIP_PORT   5060
	Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS.
#define	STANDARD_TLS_PORT   5061
	Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS.
#define	SUPPORTED   1
#define	TRUE   1
#define	UNLINK(element, head, prev)
#define	XMIT_ERROR   -2
DefaultValues Default values, set and reset in reload_config before reading configuration
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, the signalling will be incorrect. 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_ALLOW_EXT_DOM   TRUE
#define	DEFAULT_ALLOWGUEST   TRUE
#define	DEFAULT_ALWAYSAUTHREJECT   FALSE
#define	DEFAULT_AUTOCREATEPEER   FALSE
#define	DEFAULT_CALLCOUNTER   FALSE
#define	DEFAULT_CALLERID   "asterisk"
#define	DEFAULT_CALLEVENTS   FALSE
#define	DEFAULT_COMPACTHEADERS   FALSE
#define	DEFAULT_CONTEXT   "default"
#define	DEFAULT_COS_AUDIO   5
#define	DEFAULT_COS_SIP   4
#define	DEFAULT_COS_TEXT   5
#define	DEFAULT_COS_VIDEO   6
#define	DEFAULT_MATCHEXTERNIPLOCALLY   FALSE
#define	DEFAULT_MAX_CALL_BITRATE   (384)
#define	DEFAULT_MOHINTERPRET   "default"
#define	DEFAULT_MOHSUGGEST   ""
#define	DEFAULT_NOTIFYCID   DISABLED
#define	DEFAULT_NOTIFYMIME   "application/simple-message-summary"
#define	DEFAULT_NOTIFYRINGING   TRUE
#define	DEFAULT_PEDANTIC   FALSE
#define	DEFAULT_QUALIFY   FALSE
#define	DEFAULT_REALM   "asterisk"
#define	DEFAULT_REGEXTENONQUALIFY   FALSE
#define	DEFAULT_RTPKEEPALIVE   0
#define	DEFAULT_SDPOWNER   "root"
#define	DEFAULT_SDPSESSION   "Asterisk PBX"
#define	DEFAULT_SRVLOOKUP   TRUE
#define	DEFAULT_T1MIN   100
#define	DEFAULT_TOS_AUDIO   0
#define	DEFAULT_TOS_SIP   0
#define	DEFAULT_TOS_TEXT   0
#define	DEFAULT_TOS_VIDEO   0
#define	DEFAULT_USERAGENT   "Asterisk PBX"
#define	DEFAULT_VMEXTEN   "asterisk"
SIPflags
Various flags for the flags field in the pvt structure Trying to sort these up (one or more of the following): D: Dialog P: Peer/user G: Global flag When flags are used by multiple structures, it is important that they have a common layout so it is easy to copy them.
#define	SIP_CALL_LIMIT   (1 << 7)
#define	SIP_DEFER_BYE_ON_TRANSFER   (1 << 10)
#define	SIP_DIRECT_MEDIA   (1 << 20)
#define	SIP_DIRECT_MEDIA_NAT   (2 << 20)
#define	SIP_DTMF   (7 << 15)
#define	SIP_DTMF_AUTO   (3 << 15)
#define	SIP_DTMF_INBAND   (1 << 15)
#define	SIP_DTMF_INFO   (2 << 15)
#define	SIP_DTMF_RFC2833   (0 << 15)
#define	SIP_DTMF_SHORTINFO   (4 << 15)
#define	SIP_FLAGS_TO_COPY
	Flags to copy from peer/user to dialog.
#define	SIP_G726_NONSTANDARD   (1 << 31)
#define	SIP_GOTREFER   (1 << 6)
#define	SIP_INC_COUNT   (1 << 8)
#define	SIP_INC_RINGING   (1 << 9)
#define	SIP_INSECURE   (3 << 23)
#define	SIP_INSECURE_INVITE   (1 << 24)
#define	SIP_INSECURE_NONE   (0 << 23)
#define	SIP_INSECURE_PORT   (1 << 23)
#define	SIP_NAT   (3 << 18)
#define	SIP_NAT_ALWAYS   (3 << 18)
#define	SIP_NAT_NEVER   (0 << 18)
#define	SIP_NAT_RFC3581   (1 << 18)
#define	SIP_NAT_ROUTE   (2 << 18)
#define	SIP_NEEDREINVITE   (1 << 4)
#define	SIP_OUTGOING   (1 << 0)
#define	SIP_PENDINGBYE   (1 << 5)
#define	SIP_PROG_INBAND   (3 << 25)
#define	SIP_PROG_INBAND_NEVER   (0 << 25)
#define	SIP_PROG_INBAND_NO   (1 << 25)
#define	SIP_PROG_INBAND_YES   (2 << 25)
#define	SIP_PROGRESS_SENT   (1 << 3)
#define	SIP_PROMISCREDIR   (1 << 11)
#define	SIP_REINVITE   (7 << 20)
#define	SIP_REINVITE_NONE   (0 << 20)
#define	SIP_REINVITE_UPDATE   (4 << 20)
#define	SIP_RINGING   (1 << 2)
#define	SIP_SENDRPID   (1 << 29)
#define	SIP_TRUSTRPID   (1 << 12)
#define	SIP_USECLIENTCODE   (1 << 14)
#define	SIP_USEREQPHONE   (1 << 13)
SIPflags2
a second page of flags (for flags[1]
#define	SIP_PAGE2_ALLOWOVERLAP   (1 << 17)
#define	SIP_PAGE2_ALLOWSUBSCRIBE   (1 << 16)
#define	SIP_PAGE2_BUGGY_MWI   (1 << 26)
#define	SIP_PAGE2_CALL_ONHOLD   (3 << 23)
#define	SIP_PAGE2_CALL_ONHOLD_ACTIVE   (1 << 23)
#define	SIP_PAGE2_CALL_ONHOLD_INACTIVE   (3 << 23)
#define	SIP_PAGE2_CALL_ONHOLD_ONEDIR   (2 << 23)
#define	SIP_PAGE2_DIALOG_ESTABLISHED   (1 << 27)
#define	SIP_PAGE2_FAX_DETECT   (3 << 28)
#define	SIP_PAGE2_FAX_DETECT_BOTH   (3 << 28)
#define	SIP_PAGE2_FAX_DETECT_CNG   (1 << 28)
#define	SIP_PAGE2_FAX_DETECT_T38   (2 << 28)
#define	SIP_PAGE2_FLAGS_TO_COPY
#define	SIP_PAGE2_FORWARD_LOOP_DETECTED   (1 << 8)
#define	SIP_PAGE2_HAVEPEERCONTEXT   (1 << 3)
#define	SIP_PAGE2_IGNORESDPVERSION   (1 << 19)
#define	SIP_PAGE2_REGISTERTRYING   (1 << 29)
#define	SIP_PAGE2_RFC2833_COMPENSATE   (1 << 25)
#define	SIP_PAGE2_RPORT_PRESENT   (1 << 10)
#define	SIP_PAGE2_RTAUTOCLEAR   (1 << 2)
#define	SIP_PAGE2_RTCACHEFRIENDS   (1 << 0)
#define	SIP_PAGE2_STATECHANGEQUEUE   (1 << 9)
#define	SIP_PAGE2_SUBSCRIBEMWIONLY   (1 << 18)
#define	SIP_PAGE2_T38SUPPORT   (3 << 20)
#define	SIP_PAGE2_T38SUPPORT_UDPTL   (1 << 20)
#define	SIP_PAGE2_T38SUPPORT_UDPTL_FEC   (2 << 20)
#define	SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY   (3 << 20)
#define	SIP_PAGE2_TEXTSUPPORT   (1 << 15)
#define	SIP_PAGE2_UDPTL_DESTINATION   (1 << 30)
#define	SIP_PAGE2_VIDEOSUPPORT   (1 << 14)
#define	SIP_PAGE2_VIDEOSUPPORT_ALWAYS   (1 << 31)
Enumerations
enum	can_create_dialog { CAN_NOT_CREATE_DIALOG, CAN_CREATE_DIALOG, CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
	States whether a SIP message can create a dialog in Asterisk. More...
enum	check_auth_result {   AUTH_DONT_KNOW = -100, 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,   AUTH_PEER_NOT_DYNAMIC = -6, AUTH_ACL_FAILED = -7, AUTH_BAD_TRANSPORT = -8 }
	Authentication result from check_auth* functions. More...
enum	invitestates {   INV_NONE = 0, INV_CALLING = 1, INV_PROCEEDING = 2, INV_EARLY_MEDIA = 3,   INV_COMPLETED = 4, INV_CONFIRMED = 5, INV_TERMINATED = 6, INV_CANCELLED = 7 }
	States for the INVITE transaction, not the dialog. More...
enum	media_type { SDP_AUDIO, SDP_VIDEO, SDP_IMAGE, SDP_TEXT }
enum	notifycid_setting { DISABLED = 0, ENABLED = 1, IGNORE_CONTEXT = 2 }
	Settings for the 'notifycid' option, see sip.conf.sample for details. More...
enum	parse_register_result { PARSE_REGISTER_DENIED, PARSE_REGISTER_FAILED, PARSE_REGISTER_UPDATE, PARSE_REGISTER_QUERY }
	Results from the parse_register() function. More...
enum	peer_unlink_flag_t { SIP_PEERS_MARKED, SIP_PEERS_ALL }
enum	referstatus {   REFER_IDLE, REFER_SENT, REFER_RECEIVED, REFER_CONFIRMED,   REFER_ACCEPTED, REFER_RINGING, REFER_200OK, REFER_FAILED,   REFER_NOAUTH }
	Parameters to know status of transfer. More...
enum	sip_auth_type { PROXY_AUTH = 407, WWW_AUTH = 401 }
	Authentication types - proxy or www authentication. More...
enum	sip_debug_e { sip_debug_none = 0, sip_debug_config = 1, sip_debug_console = 2 }
	debugging state We store separately the debugging requests from the config file and requests from the CLI. Debugging is enabled if either is set (which means that if sipdebug is set in the config file, we can only turn it off by reloading the config). More...
enum	sip_peer_type { SIP_TYPE_PEER = (1 << 0), SIP_TYPE_USER = (1 << 1) }
enum	sip_result { AST_SUCCESS = 0, AST_FAILURE = -1 }
	The result of a lot of functions. More...
enum	sip_tcptls_alert { TCPTLS_ALERT_DATA, TCPTLS_ALERT_STOP }
enum	sip_transport { SIP_TRANSPORT_UDP = 1, SIP_TRANSPORT_TCP = 1 << 1, SIP_TRANSPORT_TLS = 1 << 2 }
	Define some implemented SIP transports. More...
enum	sipmethod {   SIP_UNKNOWN, SIP_RESPONSE, SIP_REGISTER, SIP_OPTIONS,   SIP_NOTIFY, SIP_INVITE, SIP_ACK, SIP_PRACK,   SIP_BYE, SIP_REFER, SIP_SUBSCRIBE, SIP_MESSAGE,   SIP_UPDATE, SIP_INFO, SIP_CANCEL, SIP_PUBLISH,   SIP_PING }
	SIP Request methods known by Asterisk. More...
enum	sipregistrystate {   REG_STATE_UNREGISTERED = 0, REG_STATE_REGSENT, REG_STATE_AUTHSENT, REG_STATE_REGISTERED,   REG_STATE_REJECTED, REG_STATE_TIMEOUT, REG_STATE_NOAUTH, REG_STATE_FAILED }
	States for outbound registrations (with register= lines in sip.conf. More...
enum	st_mode { SESSION_TIMER_MODE_INVALID = 0, SESSION_TIMER_MODE_ACCEPT, SESSION_TIMER_MODE_ORIGINATE, SESSION_TIMER_MODE_REFUSE }
	Modes in which Asterisk can be configured to run SIP Session-Timers. More...
enum	st_refresher { SESSION_TIMER_REFRESHER_AUTO, SESSION_TIMER_REFRESHER_UAC, SESSION_TIMER_REFRESHER_UAS }
	The entity playing the refresher role for Session-Timers. More...
enum	subscriptiontype {   NONE = 0, XPIDF_XML, DIALOG_INFO_XML, CPIM_PIDF_XML,   PIDF_XML, MWI_NOTIFICATION }
	Type of subscription, based on the packages we do support, see subscription_types. More...
enum	t38_action_flag { SDP_T38_NONE = 0, SDP_T38_INITIATE, SDP_T38_ACCEPT }
enum	t38state { T38_DISABLED = 0, T38_LOCAL_REINVITE, T38_PEER_REINVITE, T38_ENABLED }
	T38 States for a call. More...
enum	transfermodes { TRANSFER_OPENFORALL, TRANSFER_CLOSED }
	Authorization scheme for call transfers. More...
enum	xmittype { XMIT_CRITICAL = 2, XMIT_RELIABLE = 1, XMIT_UNRELIABLE = 0 }
	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. More...
Functions
static const char *	__get_header (const struct sip_request *req, const char *name, int *start)
static void	__init_check_auth_buf (void)
static void	__init_ts_temp_pvt (void)
	A per-thread temporary pvt structure.
static void	__reg_module (void)
static int	__set_address_from_contact (const char *fullcontact, struct sockaddr_in *sin, int tcp)
static int	__sip_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod)
	Acknowledges receipt of a packet and stops retransmission called with p locked.
static int	__sip_autodestruct (const void *data)
	Kill a SIP dialog (called only by the scheduler) The scheduler has a reference to this dialog when p->autokillid != -1, and we are called using that reference. So if the event is not rescheduled, we need to call dialog_unref().
static void	__sip_destroy (struct sip_pvt *p, int lockowner, int lockdialoglist)
	Execute destruction of SIP dialog structure, release memory.
static int	__sip_do_register (struct sip_registry *r)
	Register with SIP proxy.
static void	__sip_pretend_ack (struct sip_pvt *p)
	Pretend to ack all packets called with p locked.
static int	__sip_reliable_xmit (struct sip_pvt *p, int seqno, int resp, struct ast_str *data, int len, int fatal, int sipmethod)
	Transmit packet with retransmits.
static int	__sip_semi_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod)
	Acks receipt of packet, keep it around (used for provisional responses).
static int	__sip_subscribe_mwi_do (struct sip_subscription_mwi *mwi)
	Actually setup an MWI subscription or resubscribe.
static int	__sip_xmit (struct sip_pvt *p, struct ast_str *data, int len)
	Transmit SIP message Sends a SIP request or response on a given socket (in the pvt) Called by retrans_pkt, send_request, send_response and __sip_reliable_xmit.
static int	__transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Base transmit response function.
static void	__unreg_module (void)
static char *	_sip_qualify_peer (int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Send qualify message to peer from cli or manager. Mostly for debugging.
static char *	_sip_show_peer (int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Show one peer in detail (main function).
static char *	_sip_show_peers (int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Execute sip show peers command.
static void *	_sip_tcp_helper_thread (struct sip_pvt *pvt, struct ast_tcptls_session_instance *tcptls_session)
	SIP TCP thread management function This function reads from the socket, parses the packet into a request.
static int	acf_channel_read (struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
static void	add_blank (struct sip_request *req)
	add a blank line if no body
static void	add_codec_to_sdp (const struct sip_pvt *p, int codec, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add codec offer to SDP offer/answer body in INVITE or 200 OK.
static int	add_content (struct sip_request *req, const char *line)
	Add content (not header) to SIP message.
static int	add_digit (struct sip_request *req, char digit, unsigned int duration, int mode)
	Add DTMF INFO tone to sip message Mode = 0 for application/dtmf-relay (Cisco) 1 for application/dtmf.
static int	add_header (struct sip_request *req, const char *var, const char *value)
	Add header to SIP message.
static void	add_noncodec_to_sdp (const struct sip_pvt *p, int format, struct ast_str **m_buf, struct ast_str **a_buf, int debug)
	Add RFC 2833 DTMF offer to SDP.
static void	add_peer_mailboxes (struct sip_peer *peer, const char *value)
static void	add_peer_mwi_subs (struct sip_peer *peer)
static struct sip_auth *	add_realm_authentication (struct sip_auth *authlist, const char *configuration, int lineno)
	Add realm authentication in list.
static void	add_route (struct sip_request *req, struct sip_route *route)
	Add route header into request per learned route.
static enum sip_result	add_sdp (struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
	Add Session Description Protocol message.
static int	add_sip_domain (const char *domain, const enum domain_mode mode, const char *context)
	Add SIP domain to list of domains we are responsible for.
static int	add_supported_header (struct sip_pvt *pvt, struct sip_request *req)
	Add "Supported" header to sip message. Since some options may be disabled in the config, the sip_pvt must be inspected to determine what is supported for this dialog.
static void	add_tcodec_to_sdp (const struct sip_pvt *p, int codec, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add text codec offer to SDP offer/answer body in INVITE or 200 OK.
static int	add_text (struct sip_request *req, const char *text)
	Add text body to SIP message.
static struct ast_variable *	add_var (const char *buf, struct ast_variable *list)
	implement the setvar config line
static void	add_vcodec_to_sdp (const struct sip_pvt *p, int codec, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add video codec offer to SDP offer/answer body in INVITE or 200 OK.
static int	add_vidupdate (struct sip_request *req)
	add XML encoded media control with update
static int	addr_is_multicast (struct in_addr *addr)
	Check if an ip is an multicast IP. addr the address to check.
static void	append_date (struct sip_request *req)
	Append date to SIP message.
static void	append_history_full (struct sip_pvt *p, const char *fmt,...)
	Append to SIP dialog history with arg list.
static void	append_history_va (struct sip_pvt *p, const char *fmt, va_list ap)
	Append to SIP dialog history with arg list.
static void	ast_quiet_chan (struct ast_channel *chan)
	Turn off generator data XXX Does this function belong in the SIP channel?
static void	ast_sip_ouraddrfor (struct in_addr *them, struct sockaddr_in *us, struct sip_pvt *p)
	NAT fix - decide which IP address to use for Asterisk server?
static int	attempt_transfer (struct sip_dual *transferer, struct sip_dual *target)
	Attempt transfer of SIP call This fix for attended transfers on a local PBX.
static void	auth_headers (enum sip_auth_type code, char **header, char **respheader)
	return the request and response heade for a 401 or 407 code
static int	auto_congest (const void *arg)
	Scheduled congestion on a call. Only called by the scheduler, must return the reference when done.
static void	build_callid_pvt (struct sip_pvt *pvt)
	Build SIP Call-ID value for a non-REGISTER transaction.
static void	build_callid_registry (struct sip_registry *reg, struct in_addr ourip, const char *fromdomain)
	Build SIP Call-ID value for a REGISTER transaction.
static void	build_contact (struct sip_pvt *p)
	Build contact header - the contact header we send out.
static struct sip_peer *	build_peer (const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime, int devstate_only)
	Build peer from configuration (file or realtime static/dynamic).
static int	build_reply_digest (struct sip_pvt *p, int method, char *digest, int digest_len)
	Build reply digest.
static void	build_route (struct sip_pvt *p, struct sip_request *req, int backwards)
	Build route list from Record-Route header.
static void	build_rpid (struct sip_pvt *p)
	Build the Remote Party-ID & From using callingpres options.
static void	build_via (struct sip_pvt *p)
	Build a Via header for a request.
static int	cb_extensionstate (char *context, char *exten, int state, void *data)
	Callback for the devicestate notification (SUBSCRIBE) support subsystem.
static void	change_hold_state (struct sip_pvt *dialog, struct sip_request *req, int holdstate, int sendonly)
	Change hold state for a call.
static void	change_t38_state (struct sip_pvt *p, int state)
	Change the T38 state on a SIP dialog.
static enum check_auth_result	check_auth (struct sip_pvt *p, struct sip_request *req, const char *username, const char *secret, const char *md5secret, int sipmethod, char *uri, enum xmittype reliable, int ignore)
	Check user authorization from peer definition Some actions, like REGISTER and INVITEs from peers require authentication (if peer have secret set).
static enum check_auth_result	check_peer_ok (struct sip_pvt *p, char *of, struct sip_request *req, int sipmethod, struct sockaddr_in *sin, struct sip_peer **authpeer, enum xmittype reliable, char *rpid_num, char *calleridname, char *uri2)
	Validate device authentication.
static void	check_pendings (struct sip_pvt *p)
	Check pending actions on SIP call.
static void	check_rtp_timeout (struct sip_pvt *dialog, time_t t)
	helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked
static int	check_sip_domain (const char *domain, char *context, size_t len)
	check_sip_domain: Check if domain part of uri is local to our server
static int	check_user (struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin)
	Find user If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced.
static enum check_auth_result	check_user_full (struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin, struct sip_peer **authpeer)
	Check if matching user or peer is defined Match user on From: user name and peer on IP/port This is used on first invite (not re-invites) and subscribe requests.
static void	check_via (struct sip_pvt *p, struct sip_request *req)
	check Via: header for hostname, port and rport request/answer
static attribute_unused void	check_via_response (struct sip_pvt *p, struct sip_request *req)
	check received= and rport= in a SIP response. If we get a response with received= and/or rport= in the Via: line, use them as 'p->ourip' (see RFC 3581 for rport, and RFC 3261 for received). Using these two fields SIP can produce the correct address and port in the SIP headers without the need for STUN. The address part is also reused for the media sessions. Note that ast_sip_ouraddrfor() still rewrites p->ourip if you specify externip/seternaddr/stunaddr.
static void	cleanup_stale_contexts (char *new, char *old)
	Destroy disused contexts between reloads Only used in reload_config so the code for regcontext doesn't get ugly.
static void	clear_peer_mailboxes (struct sip_peer *peer)
static int	clear_realm_authentication (struct sip_auth *authlist)
	Clear realm authentication list (at reload).
static void	clear_sip_domains (void)
	Clear our domain list (at reload).
static const char *	cli_yesno (int x)
	return Yes or No depending on the argument. This is used in many places in CLI command, having a function to generate this helps maintaining a consistent output (and possibly emitting the output in other languages, at some point).
static char *	complete_sip_peer (const char *word, int state, int flags2)
	Do completion on peer name.
static char *	complete_sip_registered_peer (const char *word, int state, int flags2)
	Do completion on registered peer name.
static char *	complete_sip_show_history (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show history' CLI.
static char *	complete_sip_show_peer (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show peer' CLI.
static char *	complete_sip_show_user (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show user' CLI.
static char *	complete_sip_unregister (const char *line, const char *word, int pos, int state)
	Support routine for 'sip unregister' CLI.
static char *	complete_sip_user (const char *word, int state)
	Do completion on user name.
static char *	complete_sipch (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show channel' and 'sip show history' CLI This is in charge of generating all strings that match a prefix in the given position. As many functions of this kind, each invokation has O(state) time complexity so be careful in using it.
static char *	complete_sipnotify (const char *line, const char *word, int pos, int state)
	Support routine for 'sip notify' CLI.
static int	copy_all_header (struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy all headers from one request to another.
static int	copy_header (struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy one header field from one request to another.
static void	copy_request (struct sip_request *dst, const struct sip_request *src)
	copy SIP request (mostly used to save request for responses)
static void	copy_socket_data (struct sip_socket *to_sock, const struct sip_socket *from_sock)
static struct ast_variable *	copy_vars (struct ast_variable *src)
static int	copy_via_headers (struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy SIP VIA Headers from the request to the response.
static int	create_addr (struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin, int newdialog)
	create address structure from device name Or, if peer not found, find it in the global DNS returns TRUE (-1) on failure, FALSE on success
static int	create_addr_from_peer (struct sip_pvt *dialog, struct sip_peer *peer)
	Create address structure from peer reference. This function copies data from peer to the dialog, so we don't have to look up the peer again from memory or database during the life time of the dialog.
static void	deinit_req (struct sip_request *req)
	Deinitialize SIP response/request.
static void	destroy_association (struct sip_peer *peer)
	Remove registration data from realtime database or AST/DB when registration expires.
static void	destroy_mailbox (struct sip_mailbox *mailbox)
static int	determine_firstline_parts (struct sip_request *req)
	Parse first line of incoming SIP request.
static int	dialog_cmp_cb (void *obj, void *arg, int flags)
static int	dialog_dump_func (void *userobj, void *arg, int flags)
static int	dialog_hash_cb (const void *obj, const int flags)
static int	dialog_needdestroy (void *dialogobj, void *arg, int flags)
	Match dialogs that need to be destroyed.
static struct sip_pvt *	dialog_ref (struct sip_pvt *p, char *tag)
	when we create or delete references, make sure to use these functions so we keep track of the refcounts. To simplify the code, we allow a NULL to be passed to dialog_unref().
static void *	dialog_unlink_all (struct sip_pvt *dialog, int lockowner, int lockdialoglist)
	Unlink a dialog from the dialogs container, as well as any other places that it may be currently stored.
static struct sip_pvt *	dialog_unref (struct sip_pvt *p, char *tag)
static void	display_nat_warning (const char *cat, int reason, struct ast_flags *flags)
static int	do_magic_pickup (struct ast_channel *channel, const char *extension, const char *context)
static void *	do_monitor (void *data)
	The SIP monitoring thread.
static int	do_proxy_auth (struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code, int sipmethod, int init)
	Add authentication on outbound SIP packet.
static int	do_register_auth (struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code)
	Authenticate for outbound registration.
static void	do_setnat (struct sip_pvt *p, int natflags)
	Set nat mode on the various data sockets.
static const char *	domain_mode_to_text (const enum domain_mode mode)
	Print domain mode to cli.
static const char *	dtmfmode2str (int mode)
	Convert DTMF mode to printable string.
static int	expire_register (const void *data)
	Expire registration of SIP peer.
static void	extract_uri (struct sip_pvt *p, struct sip_request *req)
	Check Contact: URI of SIP message.
static const char *	faxec2str (int faxec)
static int	finalize_content (struct sip_request *req)
	Add 'Content-Length' header and content to SIP message.
static const char *	find_alias (const char *name, const char *_default)
	Find compressed SIP alias.
static int	find_by_name (void *obj, void *arg, void *data, int flags)
static struct sip_pvt *	find_call (struct sip_request *req, struct sockaddr_in *sin, const int intended_method)
	find or create a dialog structure for an incoming SIP message. Connect incoming SIP message to current dialog or create new dialog structure Returns a reference to the sip_pvt object, remember to give it back once done. Called by handle_request_do
static int	find_call_cb (void *__pvt, void *__arg, int flags)
static int	find_calling_channel (struct ast_channel *c, void *data)
static const char *	find_closing_quote (const char *start, const char *lim)
	Locate closing quote in a string, skipping escaped quotes. optionally with a limit on the search. start must be past the first quote.
static struct sip_peer *	find_peer (const char *peer, struct sockaddr_in *sin, int realtime, int which_objects, int devstate_only, int transport)
	Locate device by name or ip address.
static struct sip_auth *	find_realm_authentication (struct sip_auth *authlist, const char *realm)
	Find authentication for a specific realm.
static int	find_sdp (struct sip_request *req)
	Determine whether a SIP message contains an SDP in its body.
static int	find_sip_method (const char *msg)
	find_sip_method: Find SIP method from header
static struct cfsubscription_types *	find_subscription_type (enum subscriptiontype subtype)
	Find subscription type in array.
static void	free_old_route (struct sip_route *route)
	Remove route from route list.
static void	free_via (struct sip_via *v)
static int	func_check_sipdomain (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	Dial plan function to check if domain is local.
static int	func_header_read (struct ast_channel *chan, const char *function, char *data, char *buf, size_t len)
	Read SIP header (dialplan function).
static int	function_sipchaninfo_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	${SIPCHANINFO()} Dialplan function - reads sip channel data
static int	function_sippeer (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	${SIPPEER()} Dialplan function - reads peer data
static char *	generate_random_string (char *buf, size_t size)
	Generate 32 byte random string for callid's etc.
static int	get_also_info (struct sip_pvt *p, struct sip_request *oreq)
	Call transfer support (old way, deprecated by the IETF).
static char *	get_body (struct sip_request *req, char *name, char delimiter)
	Get a specific line from the message body.
static char *	get_body_by_line (const char *line, const char *name, int nameLen, char delimiter)
	Reads one line of SIP message body.
static int	get_cached_mwi (struct sip_peer *peer, int *new, int *old)
	Get cached MWI info.
static char *	get_calleridname (const char *input, char *output, size_t outputsize)
	Get caller id name from SIP headers.
static int	get_destination (struct sip_pvt *p, struct sip_request *oreq)
	Find out who the call is for. We use the request uri as a destination. This code assumes authentication has been done, so that the device (peer/user) context is already set.
static const char *	get_header (const struct sip_request *req, const char *name)
	Get header from SIP request.
static char *	get_in_brackets (char *tmp)
	Pick out text in brackets from character string.
static struct ast_variable *	get_insecure_variable_from_config (struct ast_config *config)
static int	get_ip_and_port_from_sdp (struct sip_request *req, const enum media_type media, struct sockaddr_in *sin)
static int	get_msg_text (char *buf, int len, struct sip_request *req, int addnewline)
	Get text out of a SIP MESSAGE packet.
static const char *	get_name_from_variable (struct ast_variable *var, const char *newpeername)
static void	get_our_media_address (struct sip_pvt *p, int needvideo, struct sockaddr_in *sin, struct sockaddr_in *vsin, struct sockaddr_in *tsin, struct sockaddr_in *dest, struct sockaddr_in *vdest)
	Set all IP media addresses for this call.
static int	get_rdnis (struct sip_pvt *p, struct sip_request *oreq)
	Get referring dnis.
static int	get_refer_info (struct sip_pvt *transferer, struct sip_request *outgoing_req)
	Call transfer support (the REFER method) Extracts Refer headers into pvt dialog structure.
static int	get_rpid_num (const char *input, char *output, int maxlen)
	Get caller id number from Remote-Party-ID header field Returns true if number should be restricted (privacy setting found) output is set to NULL if no number found.
static const char *	get_sdp_iterate (int *start, struct sip_request *req, const char *name)
	Lookup 'name' in the SDP starting at the 'start' line. Returns the matching line, and 'start' is updated with the next line number.
static char	get_sdp_line (int *start, int stop, struct sip_request *req, const char **value)
	Fetches the next valid SDP line between the 'start' line (inclusive) and the 'stop' line (exclusive). Returns the type ('a', 'c', ...) and matching line in reference 'start' is updated with the next line number.
static struct sip_pvt *	get_sip_pvt_byid_locked (const char *callid, const char *totag, const char *fromtag)
	Lock dialog lock and find matching pvt lock.
static const char *	get_srv_protocol (enum sip_transport t)
	Return protocol string for srv dns query.
static const char *	get_srv_service (enum sip_transport t)
	Return service string for srv dns query.
static const char *	get_transport (enum sip_transport t)
	Return transport as string.
static const char *	get_transport_list (unsigned int transports)
	Return configuration of transports for a device.
static const char *	get_transport_pvt (struct sip_pvt *p)
	Return transport of dialog.
static int	get_transport_str2enum (const char *transport)
	Return int representing a bit field of transport types found in const char *transport.
static const char *	gettag (const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize)
	Get tag from packet.
static int	handle_common_options (struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v)
	Handle flag-type options common to configuration of devices - peers.
static int	handle_incoming (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock)
	Handle incoming SIP requests (methods).
static int	handle_invite_replaces (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *nounlock)
	Handle the transfer part of INVITE with a replaces: header, meaning a target pickup or an attended transfer. Used only once. XXX 'ignore' is unused.
static int	handle_request_bye (struct sip_pvt *p, struct sip_request *req)
	Handle incoming BYE request.
static int	handle_request_cancel (struct sip_pvt *p, struct sip_request *req)
	Handle incoming CANCEL request.
static int	handle_request_do (struct sip_request *req, struct sockaddr_in *sin)
	Handle incoming SIP message - request or response.
static void	handle_request_info (struct sip_pvt *p, struct sip_request *req)
	Receive SIP INFO Message.
static int	handle_request_invite (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e, int *nounlock)
	Handle incoming INVITE request.
static int	handle_request_message (struct sip_pvt *p, struct sip_request *req)
	Handle incoming MESSAGE request.
static int	handle_request_notify (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e)
	Handle incoming notifications.
static int	handle_request_options (struct sip_pvt *p, struct sip_request *req)
	Handle incoming OPTIONS request An OPTIONS request should be answered like an INVITE from the same UA, including SDP.
static int	handle_request_refer (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, int *nounlock)
static int	handle_request_register (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e)
	Handle incoming REGISTER request.
static int	handle_request_subscribe (struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e)
	Handle incoming SUBSCRIBE request.
static void	handle_response (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
	Handle SIP response in dialogue.
static void	handle_response_invite (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
	Handle SIP response to INVITE dialogue.
static void	handle_response_notify (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
static void	handle_response_peerpoke (struct sip_pvt *p, int resp, struct sip_request *req)
	Handle qualification responses (OPTIONS).
static void	handle_response_refer (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
static int	handle_response_register (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
	Handle responses on REGISTER to services.
static void	handle_response_subscribe (struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
static int	handle_t38_options (struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v, int *maxdatagram)
	Handle T.38 configuration options common to users and peers.
static const char *	hangup_cause2sip (int cause)
	Convert Asterisk hangup causes to SIP codes.
static int	hangup_sip2cause (int cause)
	Convert SIP hangup causes to Asterisk hangup causes.
static int	init_req (struct sip_request *req, int sipmethod, const char *recip)
	Initialize SIP request.
static int	init_resp (struct sip_request *resp, const char *msg)
	Initialize SIP response, based on SIP request.
static void	initialize_initreq (struct sip_pvt *p, struct sip_request *req)
	Initialize the initital request packet in the pvt structure. This packet is used for creating replies and future requests in a dialog.
static void	initreqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod)
	Initiate new SIP request to peer/user.
static const char *	insecure2str (int mode)
	Convert Insecure setting to printable string.
static int	interpret_t38_parameters (struct sip_pvt *p, const struct ast_control_t38_parameters *parameters)
	Helper function which updates T.38 capability information and triggers a reinvite.
static void	list_route (struct sip_route *route)
	List all routes - mostly for debugging.
static int	load_module (void)
	PBX load module - initialization.
static int	local_attended_transfer (struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno)
	Find all call legs and bridge transferee with target called from handle_request_refer.
static int	lws2sws (char *msgbuf, int len)
	Parse multiline SIP headers into one header This is enabled if pedanticsipchecking is enabled.
static void	make_our_tag (char *tagbuf, size_t len)
	Make our SIP dialog tag.
static int	manager_show_registry (struct mansession *s, const struct message *m)
	Show SIP registrations in the manager API.
static int	manager_sip_qualify_peer (struct mansession *s, const struct message *m)
	Qualify SIP peers in the manager API.
static int	manager_sip_show_peer (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
static int	manager_sip_show_peers (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
static int	manager_sipnotify (struct mansession *s, const struct message *m)
static int	map_s_x (const struct _map_x_s *table, const char *s, int errorvalue)
	map from a string to an integer value, case insensitive. If no match is found, return errorvalue.
static const char *	map_x_s (const struct _map_x_s *table, int x, const char *errorstring)
	map from an integer value to a string. If no match is found, return errorstring
static int	match_and_cleanup_peer_sched (void *peerobj, void *arg, int flags)
static int	method_match (enum sipmethod id, const char *name)
	returns true if 'name' (with optional trailing whitespace) matches the sip method 'id'. Strictly speaking, SIP methods are case SENSITIVE, but we do a case-insensitive comparison to be more tolerant. following Jon Postel's rule: Be gentle in what you accept, strict with what you send
static void	mwi_event_cb (const struct ast_event *event, void *userdata)
	Receive MWI events that we have subscribed to.
static const char *	nat2str (int nat)
	Convert NAT setting to text string.
static struct sip_proxy *	obproxy_get (struct sip_pvt *dialog, struct sip_peer *peer)
	Get default outbound proxy or global proxy.
static void	parse_copy (struct sip_request *dst, const struct sip_request *src)
	Copy SIP request, parse it.
static int	parse_minse (const char *p_hdrval, int *const p_interval)
	Session-Timers: Function for parsing Min-SE header.
static void	parse_moved_contact (struct sip_pvt *p, struct sip_request *req)
	Parse 302 Moved temporalily response.
static int	parse_ok_contact (struct sip_pvt *pvt, struct sip_request *req)
	Save contact header for 200 OK on INVITE.
static enum parse_register_result	parse_register_contact (struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
	Parse contact header and save registration (peer registration).
static int	parse_request (struct sip_request *req)
	Parse a SIP message.
static int	parse_session_expires (const char *p_hdrval, int *const p_interval, enum st_refresher *const p_ref)
	Session-Timers: Function for parsing Session-Expires header.
static unsigned int	parse_sip_options (struct sip_pvt *pvt, const char *supported)
	Parse supported header in incoming packet.
static int	parse_uri (char *uri, const char *scheme, char **ret_name, char **pass, char **domain, char **port, char **options, char **transport)
	* parses a URI in its components.
static struct sip_via *	parse_via (const char *header)
	Parse a Via header.
static int	peer_cmp_cb (void *obj, void *arg, int flags)
static int	peer_dump_func (void *userobj, void *arg, int flags)
static int	peer_hash_cb (const void *obj, const int flags)
static int	peer_ipcmp_cb (void *obj, void *arg, int flags)
static int	peer_iphash_cb (const void *obj, const int flags)
static void	peer_mailboxes_to_str (struct ast_str **mailbox_str, struct sip_peer *peer)
	list peer mailboxes to CLI
static int	peer_markall_func (void *device, void *arg, int flags)
static void	peer_sched_cleanup (struct sip_peer *peer)
static int	peer_status (struct sip_peer *peer, char *status, int statuslen)
int	peercomparefunc (const void *a, const void *b)
static int	port_str2int (const char *pt, unsigned int standard)
	converts ascii port to int representation. If no pt buffer is provided or the pt has errors when being converted to an int value, the port provided as the standard is used.
static void	print_codec_to_cli (int fd, struct ast_codec_pref *pref)
	Print codec list from preference to CLI/manager.
static void	print_group (int fd, ast_group_t group, int crlf)
	Print call group and pickup group.
static void	proc_422_rsp (struct sip_pvt *p, struct sip_request *rsp)
	Handle 422 response to INVITE with session-timer requested.
static int	proc_session_timer (const void *vp)
	Session-Timers: Process session refresh timeout event.
static void	process_request_queue (struct sip_pvt *p, int *recount, int *nounlock)
static int	process_sdp (struct sip_pvt *p, struct sip_request *req, int t38action)
	Process SIP SDP offer, select formats and activate RTP channels If offer is rejected, we will not change any properties of the call Return 0 on success, a negative value on errors. Must be called after find_sdp().
static int	process_sdp_a_audio (const char *a, struct sip_pvt *p, struct ast_rtp *newaudiortp, int *last_rtpmap_codec)
static int	process_sdp_a_image (const char *a, struct sip_pvt *p)
static int	process_sdp_a_sendonly (const char *a, int *sendonly)
static int	process_sdp_a_text (const char *a, struct sip_pvt *p, struct ast_rtp *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec)
static int	process_sdp_a_video (const char *a, struct sip_pvt *p, struct ast_rtp *newvideortp, int *last_rtpmap_codec)
static int	process_sdp_c (const char *c, struct ast_hostent *hp)
static int	process_sdp_o (const char *o, struct sip_pvt *p)
static int	process_via (struct sip_pvt *p, const struct sip_request *req)
	Process the Via header according to RFC 3261 section 18.2.2.
static struct sip_proxy *	proxy_allocate (char *name, char *port, int force)
	Allocate and initialize sip proxy.
static int	proxy_update (struct sip_proxy *proxy)
static void	pvt_set_needdestroy (struct sip_pvt *pvt, const char *reason)
static int	queue_request (struct sip_pvt *p, const struct sip_request *req)
static struct sip_peer *	realtime_peer (const char *newpeername, struct sockaddr_in *sin, int devstate_only)
	realtime_peer: Get peer from realtime storage Checks the "sippeers" realtime family from extconfig.conf Checks the "sipregs" realtime family from extconfig.conf if it's configured. This returns a pointer to a peer and because we use build_peer, we can rest assured that the refcount is bumped.
static void	realtime_update_peer (const char *peername, struct sockaddr_in *sin, const char *defaultuser, const char *fullcontact, const char *useragent, int expirey, int deprecated_username, int lastms)
	Update peer object in realtime storage If the Asterisk system name is set in asterisk.conf, we will use that name and store that in the "regserver" field in the sippeers table to facilitate multi-server setups.
static void	receive_message (struct sip_pvt *p, struct sip_request *req)
	Receive SIP MESSAGE method messages.
static struct sip_peer *	ref_peer (struct sip_peer *peer, char *tag)
static void	ref_proxy (struct sip_pvt *pvt, struct sip_proxy *proxy)
	maintain proper refcounts for a sip_pvt's outboundproxy
static const char *	referstatus2str (enum referstatus rstatus)
	Convert transfer status to string.
static void	reg_source_db (struct sip_peer *peer)
	Get registration details from Asterisk DB.
static void	register_peer_exten (struct sip_peer *peer, int onoff)
	Automatically add peer extension to dial plan.
static enum check_auth_result	register_verify (struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req, char *uri)
	Verify registration of user Registration is done in several steps, first a REGISTER without auth to get a challenge (nonce) then a second one with auth Registration requests are only matched with peers that are marked as "dynamic".
static struct sip_registry *	registry_addref (struct sip_registry *reg, char *tag)
	Add object reference to SIP registry.
static void *	registry_unref (struct sip_registry *reg, char *tag)
static const char *	regstate2str (enum sipregistrystate regstate)
	Convert registration state status to string.
static int	reload (void)
	Part of Asterisk module interface.
static int	reload_config (enum channelreloadreason reason)
	Re-read SIP.conf config file.
static char *	remove_uri_parameters (char *uri)
static void	replace_cid (struct sip_pvt *p, const char *rpid_num, const char *calleridname)
	helper function for check_{user|peer}_ok()
static int	reply_digest (struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len)
	reply to authentication for outbound registrations
static int	reqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch)
	Initialize a SIP request message (not the initial one in a dialog).
static int	resp_needs_contact (const char *msg, enum sipmethod method)
	Test if this response needs a contact header.
static int	respprep (struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Prepare SIP response packet.
static int	restart_monitor (void)
	Start the channel monitor thread.
static void	restart_session_timer (struct sip_pvt *p)
	Session-Timers: Restart session timer.
static int	retrans_pkt (const void *data)
	Retransmit SIP message if no answer (Called from scheduler).
static int	scheduler_process_request_queue (const void *data)
static int	send_provisional_keepalive (const void *data)
static int	send_provisional_keepalive_full (struct sip_pvt *pvt, int with_sdp)
static int	send_provisional_keepalive_with_sdp (const void *data)
static int	send_request (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
	Send SIP Request to the other part of the dialogue.
static int	send_response (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
	Transmit response on SIP request.
static int	set_address_from_contact (struct sip_pvt *pvt)
	Change the other partys IP address based on given contact.
static void	set_destination (struct sip_pvt *p, char *uri)
	Set destination from SIP URI.
static void	set_insecure_flags (struct ast_flags *flags, const char *value, int lineno)
	Parse insecure= setting in sip.conf and set flags according to setting.
static void	set_nonce_randdata (struct sip_pvt *p, int forceupdate)
	builds the sip_pvt's randdata field which is used for the nonce challenge. When forceupdate is not set, the nonce is only updated if the current one is stale. In this case, a stalenonce is one which has already received a response, if a nonce has not received a response it is not always necessary or beneficial to create a new one.
static void	set_peer_defaults (struct sip_peer *peer)
	Set peer defaults before configuring specific configurations.
static void	set_socket_transport (struct sip_socket *socket, int transport)
static void	set_t38_capabilities (struct sip_pvt *p)
	Set the global T38 capabilities on a SIP dialog structure.
static int	show_channels_cb (void *__cur, void *__arg, int flags)
	callback for show channel|subscription
static int	show_chanstats_cb (void *__cur, void *__arg, int flags)
	Callback for show_chanstats.
static int	sip_addheader (struct ast_channel *chan, void *data)
	Add a SIP header to an outbound INVITE.
static struct sip_pvt *	sip_alloc (ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, struct sip_request *req)
	Allocate sip_pvt structure, set defaults and link in the container. Returns a reference to the object so whoever uses it later must remember to release the reference.
static void	sip_alreadygone (struct sip_pvt *dialog)
	Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging.
static int	sip_answer (struct ast_channel *ast)
	sip_answer: Answer SIP call , send 200 OK on Invite Part of PBX interface
static int	sip_call (struct ast_channel *ast, char *dest, int timeout)
	Initiate SIP call from PBX used from the dial() application.
static int	sip_cancel_destroy (struct sip_pvt *p)
	Cancel destruction of SIP dialog. Be careful as this also absorbs the reference - if you call it from within the scheduler, this might be the last reference.
static int	sip_check_authtimeout (time_t start)
	Check if the authtimeout has expired.
static char *	sip_cli_notify (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Cli command to send SIP notify to peer.
static int	sip_debug_test_addr (const struct sockaddr_in *addr)
	See if we pass debug IP filter.
static int	sip_debug_test_pvt (struct sip_pvt *p)
	Test PVT for debugging output.
static struct sip_pvt *	sip_destroy (struct sip_pvt *p)
	Destroy SIP call structure. Make it return NULL so the caller can do things like foo = sip_destroy(foo); and reduce the chance of bugs due to dangling pointers.
static void	sip_destroy_fn (void *p)
static void	sip_destroy_peer (struct sip_peer *peer)
	Destroy peer object from memory.
static void	sip_destroy_peer_fn (void *peer)
static int	sip_devicestate (void *data)
	Part of PBX channel interface.
static char *	sip_do_debug (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Turn on SIP debugging (CLI command).
static char *	sip_do_debug_ip (int fd, char *arg)
	Enable SIP Debugging for a single IP.
static char *	sip_do_debug_peer (int fd, char *arg)
	Turn on SIP debugging for a given peer.
static int	sip_do_reload (enum channelreloadreason reason)
	Reload module.
static int	sip_dtmfmode (struct ast_channel *chan, void *data)
	Set the DTMFmode for an outbound SIP call (application).
static void	sip_dump_history (struct sip_pvt *dialog)
	Dump SIP history to debug log file at end of lifespan for SIP dialog.
static int	sip_fixup (struct ast_channel *oldchan, struct ast_channel *newchan)
	sip_fixup: Fix up a channel: If a channel is consumed, this is called. Basically update any ->owner links
static const char *	sip_get_callid (struct ast_channel *chan)
	Deliver SIP call ID for the call.
static int	sip_get_codec (struct ast_channel *chan)
	Return SIP UA's codec (part of the RTP interface).
static enum ast_rtp_get_result	sip_get_rtp_peer (struct ast_channel *chan, struct ast_rtp **rtp)
	Returns null if we can't reinvite audio (part of RTP interface).
static enum ast_rtp_get_result	sip_get_trtp_peer (struct ast_channel *chan, struct ast_rtp **rtp)
	Returns null if we can't reinvite text (part of RTP interface).
static struct ast_udptl *	sip_get_udptl_peer (struct ast_channel *chan)
static enum ast_rtp_get_result	sip_get_vrtp_peer (struct ast_channel *chan, struct ast_rtp **rtp)
	Returns null if we can't reinvite video (part of RTP interface).
static int	sip_hangup (struct ast_channel *ast)
	sip_hangup: Hangup SIP call Part of PBX interface, called from ast_hangup
static int	sip_indicate (struct ast_channel *ast, int condition, const void *data, size_t datalen)
	Play indication to user With SIP a lot of indications is sent as messages, letting the device play the indication - busy signal, congestion etc.
static const char *	sip_nat_mode (const struct sip_pvt *p)
	Display SIP nat mode.
static struct ast_channel *	sip_new (struct sip_pvt *i, int state, const char *title)
	Initiate a call in the SIP channel called from sip_request_call (calls from the pbx ) for outbound channels and from handle_request_invite for inbound channels.
static int	sip_park (struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno)
	Park a call using the subsystem in res_features.c This is executed in a separate thread.
static void *	sip_park_thread (void *stuff)
	Park SIP call support function Starts in a new thread, then parks the call XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the audio can't be heard before hangup.
static int	sip_parse_host (char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport)
	Small function to parse a config line for a host with a transport i.e. tls://www.google.com:8056.
static void	sip_peer_hold (struct sip_pvt *p, int hold)
	Change onhold state of a peer using a pvt structure.
static int	sip_pickup (struct ast_channel *chan)
	Pickup a call using the subsystem in features.c This is executed in a separate thread.
static void *	sip_pickup_thread (void *stuff)
	SIP pickup support function Starts in a new thread, then pickup the call.
static void	sip_poke_all_peers (void)
	Send a poke to all known peers.
static int	sip_poke_noanswer (const void *data)
	React to lack of answer to Qualify poke.
static int	sip_poke_peer (struct sip_peer *peer, int force)
	Check availability of peer, also keep NAT open.
static int	sip_poke_peer_s (const void *data)
	Poke peer (send qualify to check if peer is alive and well).
static int	sip_prepare_socket (struct sip_pvt *p)
static char *	sip_prune_realtime (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Remove temporary realtime objects from memory (CLI).
static char *	sip_qualify_peer (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Send an OPTIONS packet to a SIP peer.
static int	sip_queryoption (struct ast_channel *chan, int option, void *data, int *datalen)
	Query an option on a SIP dialog.
static struct ast_frame *	sip_read (struct ast_channel *ast)
	Read SIP RTP from channel.
static struct sockaddr_in *	sip_real_dst (const struct sip_pvt *p)
	The real destination address for a write.
static int	sip_refer_allocate (struct sip_pvt *p)
	Allocate SIP refer structure.
static int	sip_reg_timeout (const void *data)
	Registration timeout, register again Registered as a timeout handler during transmit_register(), to retransmit the packet if a reply does not come back. This is called by the scheduler so the event is not pending anymore when we are called.
static int	sip_register (const char *value, int lineno)
	Parse register=> line in sip.conf and add to registry.
static void	sip_registry_destroy (struct sip_registry *reg)
	Destroy registry object Objects created with the register= statement in static configuration.
static int	sip_reinvite_retry (const void *data)
	Reset the NEEDREINVITE flag after waiting when we get 491 on a Re-invite to avoid race conditions between asterisk servers. Called from the scheduler.
static char *	sip_reload (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Force reload of module from cli.
static int	sip_removeheader (struct ast_channel *chan, void *data)
	Remove SIP headers added previously with SipAddHeader application.
static struct ast_channel *	sip_request_call (const char *type, int format, void *data, int *cause)
	PBX interface function -build SIP pvt structure SIP calls initiated by the PBX arrive here.
static int	sip_reregister (const void *data)
	Update registration with SIP Proxy. Called from the scheduler when the previous registration expires, so we don't have to cancel the pending event. We assume the reference so the sip_registry is valid, since it is stored in the scheduled event anyways.
static struct ast_frame *	sip_rtp_read (struct ast_channel *ast, struct sip_pvt *p, int *faxdetect)
	Read RTP from network.
static void	sip_scheddestroy (struct sip_pvt *p, int ms)
	Schedule destruction of SIP dialog.
static void	sip_send_all_mwi_subscriptions (void)
	Send all MWI subscriptions.
static void	sip_send_all_registers (void)
	Send all known registrations.
static int	sip_send_mwi_to_peer (struct sip_peer *peer, const struct ast_event *event, int cache_only)
	Send message waiting indication to alert peer that they've got voicemail.
static int	sip_senddigit_begin (struct ast_channel *ast, char digit)
static int	sip_senddigit_end (struct ast_channel *ast, char digit, unsigned int duration)
	Send DTMF character on SIP channel within one call, we're able to transmit in many methods simultaneously.
static int	sip_sendhtml (struct ast_channel *chan, int subclass, const char *data, int datalen)
	Send message with Access-URL header, if this is an HTML URL only!
static int	sip_sendtext (struct ast_channel *ast, const char *text)
	Send SIP MESSAGE text within a call Called from PBX core sendtext() application.
static char *	sip_set_history (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Enable/Disable SIP History logging (CLI).
static void	sip_set_redirstr (struct sip_pvt *p, char *reason)
	Translate referring cause.
static int	sip_set_rtp_peer (struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
	Set the RTP peer for this call.
static int	sip_set_udptl_peer (struct ast_channel *chan, struct ast_udptl *udptl)
static char *	sip_show_channel (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show details of one active dialog.
static char *	sip_show_channels (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI for show channels or subscriptions. This is a new-style CLI handler so a single function contains the prototype for the function, the 'generator' to produce multiple entries in case it is required, and the actual handler for the command.
static char *	sip_show_channelstats (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	SIP show channelstats CLI (main function).
static char *	sip_show_domains (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI command to list local domains.
static char *	sip_show_history (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show history details of one dialog.
static char *	sip_show_inuse (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Command to show calls within limits set by call_limit.
static char *	sip_show_mwi (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	sip_show_objects (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	List all allocated SIP Objects (realtime or static).
static char *	sip_show_peer (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show one peer in detail.
static char *	sip_show_peers (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Show Peers command.
static char *	sip_show_registry (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show SIP Registry (registrations with other SIP proxies.
static char *	sip_show_sched (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	sip_show_settings (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	List global settings for the SIP channel.
static char *	sip_show_tcp (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show active TCP connections.
static char *	sip_show_user (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show one user in detail.
static char *	sip_show_users (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Command 'SIP Show Users'.
static int	sip_sipredirect (struct sip_pvt *p, const char *dest)
	Transfer call before connect with a 302 redirect.
static struct sip_st_dlg *	sip_st_alloc (struct sip_pvt *const p)
	Allocate Session-Timers struct w/in dialog.
static int	sip_standard_port (enum sip_transport type, int port)
	Returns the port to use for this socket.
static int	sip_subscribe_mwi (const char *value, int lineno)
	Parse mwi=> line in sip.conf and add to list.
static void	sip_subscribe_mwi_destroy (struct sip_subscription_mwi *mwi)
	Destroy MWI subscription object.
static int	sip_subscribe_mwi_do (const void *data)
	Send a subscription or resubscription for MWI.
static int	sip_t38_abort (const void *data)
	Called to deny a T38 reinvite if the core does not respond to our request.
static struct ast_tcptls_session_instance *	sip_tcp_locate (struct sockaddr_in *s)
	Find thread for TCP/TLS session (based on IP/Port.
static void *	sip_tcp_worker_fn (void *data)
	SIP TCP connection handler.
static void	sip_tcptls_client_args_destructor (void *obj)
static int	sip_tcptls_write (struct ast_tcptls_session_instance *tcptls_session, const void *buf, size_t len)
	used to indicate to a tcptls thread that data is ready to be written
static struct sip_threadinfo *	sip_threadinfo_create (struct ast_tcptls_session_instance *tcptls_session, int transport)
	creates a sip_threadinfo object and links it into the threadt table.
static void	sip_threadinfo_destructor (void *obj)
static int	sip_transfer (struct ast_channel *ast, const char *dest)
	Transfer SIP call.
static char *	sip_unregister (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Unregister (force expiration) a SIP peer in the registry via CLI.
static int	sip_uri_cmp (const char *input1, const char *input2)
static int	sip_uri_headers_cmp (const char *input1, const char *input2)
	helper routine for sip_uri_cmp
static int	sip_uri_params_cmp (const char *input1, const char *input2)
	helper routine for sip_uri_cmp to compare URI parameters
static int	sip_write (struct ast_channel *ast, struct ast_frame *frame)
	Send frame to media channel (rtp).
static int	sipsock_read (int *id, int fd, short events, void *ignore)
	Read data from SIP UDP socket.
static enum st_mode	st_get_mode (struct sip_pvt *p, int no_cached)
	Get the session-timer mode.
static enum st_refresher	st_get_refresher (struct sip_pvt *p)
	Get the entity (UAC or UAS) that's acting as the session-timer refresher.
static int	st_get_se (struct sip_pvt *p, int max)
	Get Max or Min SE (session timer expiry).
static void	start_session_timer (struct sip_pvt *p)
	Session-Timers: Start session timer.
static const char *	stmode2str (enum st_mode m)
static void	stop_media_flows (struct sip_pvt *p)
	Immediately stop RTP, VRTP and UDPTL as applicable.
static void	stop_session_timer (struct sip_pvt *p)
	Session-Timers: Stop session timer.
static int	str2dtmfmode (const char *str)
	maps a string to dtmfmode, returns -1 on error
static enum st_mode	str2stmode (const char *s)
static enum st_refresher	str2strefresher (const char *s)
static const char *	strefresher2str (enum st_refresher r)
static const char *	subscription_type2str (enum subscriptiontype subtype)
	Show subscription type in string format.
static unsigned int	t38_get_rate (enum ast_control_t38_rate rate)
	Get Max T.38 Transmission rate from T38 capabilities.
static void	tcptls_packet_destructor (void *obj)
static struct sip_peer *	temp_peer (const char *name)
	Create temporary peer (used in autocreatepeer mode).
static void	temp_pvt_cleanup (void *)
static int	temp_pvt_init (void *)
static char *	terminate_uri (char *uri)
static int	threadinfo_locate_cb (void *obj, void *arg, int flags)
static int	threadt_cmp_cb (void *obj, void *arg, int flags)
static int	threadt_hash_cb (const void *obj, const int flags)
static char *	transfermode2str (enum transfermodes mode)
	Convert transfer mode to text string.
static void	transmit_fake_auth_response (struct sip_pvt *p, int sipmethod, struct sip_request *req, enum xmittype reliable)
	Send a fake 401 Unauthorized response when the administrator wants to hide the names of local devices from fishers.
static int	transmit_info_with_digit (struct sip_pvt *p, const char digit, unsigned int duration)
	Send SIP INFO dtmf message, see Cisco documentation on cisco.com.
static int	transmit_info_with_vidupdate (struct sip_pvt *p)
	Send SIP INFO with video update request.
static int	transmit_invite (struct sip_pvt *p, int sipmethod, int sdp, int init)
	Build REFER/INVITE/OPTIONS/SUBSCRIBE message and transmit it.
static int	transmit_message_with_text (struct sip_pvt *p, const char *text)
	Transmit text with SIP MESSAGE method.
static int	transmit_notify_custom (struct sip_pvt *p, struct ast_variable *vars)
	Notify device with custom headers from sip_notify.conf.
static int	transmit_notify_with_mwi (struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten)
	Notify user of messages waiting in voicemail (RFC3842).
static int	transmit_notify_with_sipfrag (struct sip_pvt *p, int cseq, char *message, int terminate)
	Notify a transferring party of the status of transfer (RFC3515).
static int	transmit_provisional_response (struct sip_pvt *p, const char *msg, const struct sip_request *req, int with_sdp)
static int	transmit_refer (struct sip_pvt *p, const char *dest)
	Transmit SIP REFER message (initiated by the transfer() dialplan application.
static int	transmit_register (struct sip_registry *r, int sipmethod, const char *auth, const char *authheader)
	Transmit register to SIP proxy or UA auth = NULL on the initial registration (from sip_reregister()).
static int	transmit_reinvite_with_sdp (struct sip_pvt *p, int t38version, int oldsdp)
	Transmit reinvite with SDP.
static int	transmit_request (struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
	Transmit generic SIP request returns XMIT_ERROR if transmit failed with a critical error (don't retry).
static int	transmit_request_with_auth (struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
	Transmit SIP request, auth added.
static int	transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Transmit response, no retransmits.
static int	transmit_response_reliable (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Transmit response, Make sure you get an ACK This is only used for responses to INVITEs, where we need to make sure we get an ACK.
static int	transmit_response_using_temp (ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg)
	Transmit response, no retransmits, using a temporary pvt structure.
static int	transmit_response_with_allow (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Append Accept header, content length before transmitting response.
static int	transmit_response_with_auth (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *randdata, enum xmittype reliable, const char *header, int stale)
	Respond with authorization request.
static int	transmit_response_with_date (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Append date and content length before transmitting response.
static int	transmit_response_with_minse (struct sip_pvt *p, const char *msg, const struct sip_request *req, int minse_int)
	Transmit 422 response with Min-SE header (Session-Timers).
static int	transmit_response_with_sdp (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp)
	Used for 200 OK and 183 early media.
static int	transmit_response_with_t38_sdp (struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
	Used for 200 OK and 183 early media.
static int	transmit_response_with_unsupported (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported)
	Transmit response, no retransmits.
static int	transmit_state_notify (struct sip_pvt *p, int state, int full, int timeout)
	Used in the SUBSCRIBE notification subsystem (RFC3265).
static void	try_suggested_sip_codec (struct sip_pvt *p)
	Try setting codec suggested by the SIP_CODEC channel variable.
static void	unlink_all_peers_from_tables (void)
static void	unlink_marked_peers_from_tables (void)
static void	unlink_peer_from_tables (struct sip_peer *peer)
static void	unlink_peers_from_tables (peer_unlink_flag_t flag)
static int	unload_module (void)
	PBX unload module API.
static void *	unref_peer (struct sip_peer *peer, char *tag)
static int	update_call_counter (struct sip_pvt *fup, int event)
	update_call_counter: Handle call_limit for SIP devices Setting a call-limit will cause calls above the limit not to be accepted.
static void	update_peer (struct sip_peer *p, int expire)
	Update peer data in database (if used).
static void	update_provisional_keepalive (struct sip_pvt *pvt, int with_sdp)
Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_DEFAULT , .description = "Session Initiation Protocol (SIP)" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = AST_BUILDOPT_SUM, .load = load_module, .unload = unload_module, .reload = reload, }
static char *	app_dtmfmode = "SIPDtmfMode"
static char *	app_sipaddheader = "SIPAddHeader"
static char *	app_sipremoveheader = "SIPRemoveHeader"
static struct ast_module_info *	ast_module_info = &__mod_info
static struct sip_auth *	authl = NULL
	Authentication list for realm authentication.
static int	authlimit = DEFAULT_AUTHLIMIT
static int	authtimeout = DEFAULT_AUTHTIMEOUT
static struct sockaddr_in	bindaddr
static unsigned int	chan_idx
static struct ast_threadstorage	check_auth_buf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_check_auth_buf , .custom_init = NULL , }
static struct ast_custom_function	checksipdomain_function
static struct ast_cli_entry	cli_sip []
	SIP Cli commands definition.
static const char	config [] = "sip.conf"
static struct sockaddr_in	debugaddr
static int	default_expiry = DEFAULT_DEFAULT_EXPIRY
static struct ast_jb_conf	default_jbconf
	Global jitterbuffer configuration - by default, jb is disabled.
static struct ast_tls_config	default_tls_cfg
	Default TLS connection configuration.
struct ao2_container *	dialogs
	Here we implement the container for dialogs (sip_pvt), defining generic wrapper functions to ease the transition from the current implementation (a single linked list) to a different container. In addition to a reference to the container, we need functions to lock/unlock the container and individual items, and functions to add/remove references to the individual items.
static struct _map_x_s	dtmfstr []
	mapping between dtmf flags and strings
static time_t	externexpire
static char	externhost [MAXHOSTNAMELEN]
static struct sockaddr_in	externip
	our external IP address/port for SIP sessions. externip.sin_addr is only set when we know we might be behind a NAT, and this is done using a variety of (mutually exclusive) ways from the config file:
static int	externrefresh = 10
static struct _map_x_s	faxecmodes []
static struct ast_ha *	global_contact_ha = NULL
	Global list of addresses dynamic peers are not allowed to use.
static struct ast_jb_conf	global_jbconf
static const int	HASH_DIALOG_SIZE = 563
static const int	HASH_PEER_SIZE = 563
static struct _map_x_s	insecurestr []
static struct sockaddr_in	internip
	our (internal) default address/port to put in SIP/SDP messages internip is initialized picking a suitable address from one of the interfaces, and the same port number we bind to. It is used as the default address/port in SIP messages, and as the default address (but not port) in SDP messages.
static struct invstate2stringtable	invitestate2string []
	Readable descriptions of device states.
static struct ast_ha *	localaddr
	List of local networks We store "localnet" addresses from the config file into an access list, marked as 'DENY', so the call to ast_apply_ha() will return AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local' (i.e. presumably public) addresses.
static char	mandescr_show_peer []
static char	mandescr_show_peers []
static char	mandescr_show_registry []
	Manager Action SIPShowRegistry description.
static char	mandescr_sipnotify []
static int	max_expiry = DEFAULT_MAX_EXPIRY
static int	min_expiry = DEFAULT_MIN_EXPIRY
static int	mwi_expiry = DEFAULT_MWI_EXPIRY
static struct _map_x_s	natmodes []
static const char	notify_config [] = "sip_notify.conf"
static struct ast_config *	notify_types = NULL
static int	ourport_tcp
static int	ourport_tls
struct ao2_container *	peers
	The peer list: Users, Peers and Friends.
struct ao2_container *	peers_by_ip
static struct _map_x_s	referstatusstrings []
static struct ast_register_list	regl
	The register list: Other SIP proxies we register with and place calls to.
static struct _map_x_s	regstatestrings []
static struct ast_custom_function	sip_header_function
static struct cfsip_methods	sip_methods []
	The core structure to setup dialogs. We parse incoming messages by using structure and then route the messages according to the type.
static struct cfsip_options	sip_options []
	List of well-known SIP options. If we get this in a require, we should check the list and answer accordingly.
static struct ast_rtp_protocol	sip_rtp
	Interface structure with callbacks used to connect to RTP module.
static struct ast_tcptls_session_args	sip_tcp_desc
	The TCP server definition.
static struct ast_channel_tech	sip_tech
	Definition of this channel for PBX channel registration.
static struct ast_channel_tech	sip_tech_info
	This version of the sip channel tech has no send_digit_begin callback so that the core knows that the channel does not want DTMF BEGIN frames. The struct is initialized just before registering the channel driver, and is for use with channels using SIP INFO DTMF.
static struct ast_tls_config	sip_tls_cfg
	Working TLS connection configuration.
static struct ast_tcptls_session_args	sip_tls_desc
	The TCP/TLS server definition.
static struct ast_udptl_protocol	sip_udptl
	Interface structure with callbacks used to connect to UDPTL module.
static struct ast_custom_function	sipchaninfo_function
	Structure to declare a dialplan function: SIPCHANINFO.
static enum sip_debug_e	sipdebug
static int	sipdebug_text
	extra debugging for 'text' related events. At the moment this is set together with sip_debug_console.
static struct ast_custom_function	sippeer_function
	Structure to declare a dialplan function: SIPPEER.
static int	sipsock = -1
	Main socket for UDP SIP communication.
static struct _map_x_s	stmodes []
	Report Peer status in character string.
static struct _map_x_s	strefreshers []
static struct sockaddr_in	stunaddr
static struct ast_subscription_mwi_list	submwil
	The MWI subscription list.
static struct cfsubscription_types	subscription_types []
	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.
static struct ao2_container *	threadt
static struct ast_threadstorage	ts_temp_pvt = { .once = PTHREAD_ONCE_INIT , .key_init = __init_ts_temp_pvt , .custom_init = temp_pvt_init , }
static int	unauth_sessions = 0
DefaultSettings
Default setttings are used as a channel setting and as a default when configuring devices
static char	default_callerid [AST_MAX_EXTENSION]
static char	default_fromdomain [AST_MAX_EXTENSION]
static char	default_language [MAX_LANGUAGE]
static int	default_maxcallbitrate
static char	default_mohinterpret [MAX_MUSICCLASS]
static char	default_mohsuggest [MAX_MUSICCLASS]
static char	default_notifymime [AST_MAX_EXTENSION]
static char	default_parkinglot [AST_MAX_CONTEXT]
static struct ast_codec_pref	default_prefs
static unsigned int	default_primary_transport
static int	default_qualify
static unsigned int	default_transports
static char	default_vmexten [AST_MAX_EXTENSION]
GlobalSettings
Global settings apply to the channel (often settings you can change in the general section of sip.conf
static int	dumphistory
static int	global_authfailureevents
static int	global_autoframing
static int	global_callcounter
static int	global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263
	Codecs that we support by default:.
static unsigned int	global_cos_audio
static unsigned int	global_cos_sip
static unsigned int	global_cos_text
static unsigned int	global_cos_video
static int	global_dynamic_exclude_static = 0
static int	global_match_auth_username
static int	global_max_se
static int	global_min_se
static int	global_prematuremediafilter
static int	global_qualify_gap
static int	global_qualify_peers
static int	global_qualifyfreq
static int	global_reg_timeout
static int	global_regattempts_max
static char	global_regcontext [AST_MAX_CONTEXT]
static int	global_relaxdtmf
static int	global_rtpholdtimeout
static int	global_rtpkeepalive
static int	global_rtptimeout
static char	global_sdpowner [AST_MAX_EXTENSION]
static char	global_sdpsession [AST_MAX_EXTENSION]
static int	global_shrinkcallerid
static enum st_mode	global_st_mode
static enum st_refresher	global_st_refresher
static int	global_t1
static int	global_t1min
static int	global_timer_b
static unsigned int	global_tos_audio
static unsigned int	global_tos_sip
static unsigned int	global_tos_text
static unsigned int	global_tos_video
static char	global_useragent [AST_MAX_EXTENSION]
static int	recordhistory
static struct sip_settings	sip_cfg

---

Detailed Description

Implementation of Session Initiation Protocol.

Author:

Mark Spencer <markster@digium.com>

See Also:

• CREDITS

Implementation of RFC 3261 - without S/MIME, and experimental TCP and TLS support Configuration file sip.conf

********** IMPORTANT *

Note:

TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration settings, dialplan commands and dialplans apps/functions See SIP TCP and TLS support

******** General TODO:s

Todo:

Better support of forking

Todo:

VIA branch tag transaction checking

Todo:

Transaction support

******** Wishlist: Improvements

• Support of SIP domains for devices, so that we match on username in the From: header
• Connect registrations with a specific device on the incoming call. It's not done automatically in Asterisk

Overview of the handling of SIP sessions

The SIP channel handles several types of SIP sessions, or dialogs, not all of them being "telephone calls".

• Incoming calls that will be sent to the PBX core
• Outgoing calls, generated by the PBX
• SIP subscriptions and notifications of states and voicemail messages
• SIP registrations, both inbound and outbound
• SIP peer management (peerpoke, OPTIONS)
• SIP text messages

In the SIP channel, there's a list of active SIP dialogs, which includes all of these when they are active. "sip show channels" in the CLI will show most of these, excluding subscriptions which are shown by "sip show subscriptions"

incoming packets

Incoming packets are received in the monitoring thread, then handled by sipsock_read() for udp only. In tcp, packets are read by the tcp_helper thread. sipsock_read() function parses the packet and matches an existing dialog or starts a new SIP dialog.

sipsock_read sends the packet to handle_incoming(), that parses a bit more. If it is a response to an outbound request, the packet is sent to handle_response(). If it is a request, handle_incoming() sends it to one of a list of functions depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc sipsock_read locks the ast_channel if it exists (an active call) and unlocks it after we have processed the SIP message.

A new INVITE is sent to handle_request_invite(), that will end up starting a new channel in the PBX, the new channel after that executing in a separate channel thread. This is an incoming "call". When the call is answered, either by a bridged channel or the PBX itself the sip_answer() function is called.

The actual media - Video or Audio - is mostly handled by the RTP subsystem in rtp.c

Outbound calls

Outbound calls are set up by the PBX through the sip_request_call() function. After that, they are activated by sip_call().

Hanging up

The PBX issues a hangup on both incoming and outgoing calls through the sip_hangup() function

Definition in file chan_sip.c.

---

Define Documentation

#define ALLOWED_METHODS   "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO"

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.

Definition at line 955 of file chan_sip.c.

Referenced by respprep(), transmit_invite(), transmit_notify_with_sipfrag(), transmit_refer(), and transmit_reinvite_with_sdp().

#define append_history	(	p,
		event,
		fmt,
		args...		)	append_history_full(p, "%-15s " fmt, event, ## args)

Append to SIP dialog history.

Returns:

Always returns 0

Definition at line 2507 of file chan_sip.c.

Referenced by __sip_autodestruct(), __sip_reliable_xmit(), auto_congest(), build_reply_digest(), cb_extensionstate(), change_hold_state(), check_auth(), do_register_auth(), handle_invite_replaces(), handle_request_bye(), handle_request_do(), handle_request_invite(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), handle_response(), handle_response_invite(), local_attended_transfer(), obproxy_get(), pvt_set_needdestroy(), retrans_pkt(), send_request(), send_response(), sip_cancel_destroy(), sip_fixup(), sip_hangup(), sip_new(), sip_park_thread(), sip_reregister(), sip_scheddestroy(), sip_set_rtp_peer(), transmit_register(), transmit_reinvite_with_sdp(), and transmit_response_with_auth().

#define CALLERID_UNKNOWN   "Anonymous"

Definition at line 528 of file chan_sip.c.

Referenced by initreqprep().

#define CHECK_AUTH_BUF_INITLEN   256

Definition at line 13179 of file chan_sip.c.

Referenced by check_auth(), and transmit_fake_auth_response().

#define check_request_transport	(	peer,
		tmpl		)

generic function for determining if a correct transport is being used to contact a peer

this is done as a macro so that the "tmpl" var can be passed either a sip_request or a sip_peer

Definition at line 2414 of file chan_sip.c.

Referenced by create_addr_from_peer(), and register_verify().

#define DEC_CALL_LIMIT   0

Definition at line 1165 of file chan_sip.c.

Referenced by __sip_destroy(), handle_request_cancel(), handle_request_invite(), handle_response_invite(), sip_hangup(), and update_call_counter().

#define DEC_CALL_RINGING   2

Definition at line 1167 of file chan_sip.c.

Referenced by handle_response_invite(), and update_call_counter().

#define DEFAULT_ALLOW_EXT_DOM   TRUE

Allow external domains

Definition at line 997 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_ALLOWGUEST   TRUE

Definition at line 984 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_ALWAYSAUTHREJECT   FALSE

Don't reject authentication requests always

Definition at line 1006 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_AUTHLIMIT   100

Definition at line 503 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_AUTHTIMEOUT   30

Definition at line 504 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_AUTOCREATEPEER   FALSE

Don't create peers automagically

Definition at line 1002 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_CALLCOUNTER   FALSE

Definition at line 986 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_CALLERID   "asterisk"

Default caller ID

Definition at line 982 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_CALLEVENTS   FALSE

Extra manager SIP call events

Definition at line 1005 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_COMPACTHEADERS   FALSE

Send compact (one-character) SIP headers. Default off

Definition at line 988 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_CONTEXT   "default"

The default context for [general] section as well as devices

Definition at line 978 of file chan_sip.c.

#define DEFAULT_COS_AUDIO   5

Level 2 class of service for audio media

Definition at line 994 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_COS_SIP   4

Level 2 class of service for SIP signalling

Definition at line 993 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_COS_TEXT   5

Level 2 class of service for text media (T.140)

Definition at line 996 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_COS_VIDEO   6

Level 2 class of service for video media

Definition at line 995 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_DEFAULT_EXPIRY   120

Definition at line 497 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_EXPIRY   900

Expire slowly

Definition at line 517 of file chan_sip.c.

#define DEFAULT_FREQ_NOTOK   10 * 1000

Qualification: How often to check, if the host is down...

Definition at line 533 of file chan_sip.c.

#define DEFAULT_MATCHEXTERNIPLOCALLY   FALSE

Match extern IP locally default setting

Definition at line 1003 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_MAX_CALL_BITRATE   (384)

Max bitrate for video

Definition at line 1009 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_MAX_EXPIRY   3600

Definition at line 499 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_MAX_FORWARDS   "70"

Definition at line 502 of file chan_sip.c.

Referenced by initreqprep(), reqprep(), and transmit_register().

#define DEFAULT_MAX_SE   1800

Session-Timer Default Session-Expires period (RFC 4028)

Definition at line 552 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_MAXMS   2000

Qualification: Must be faster than 2 seconds by default

Definition at line 531 of file chan_sip.c.

#define DEFAULT_MIN_EXPIRY   60

Definition at line 498 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_MIN_SE   90

Session-Timer Default Min-SE period (RFC 4028)

Definition at line 553 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_MOHINTERPRET   "default"

The default music class

Definition at line 979 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_MOHSUGGEST   ""

Definition at line 980 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_MWI_EXPIRY   3600

Definition at line 500 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_NOTIFYCID   DISABLED

Include CID with ringing notifications

Definition at line 1000 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_NOTIFYMIME   "application/simple-message-summary"

Definition at line 983 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_NOTIFYRINGING   TRUE

Notify devicestate system on ringing state

Definition at line 999 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_PEDANTIC   FALSE

Avoid following SIP standards for dialog matching

Definition at line 1001 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_QUALIFY   FALSE

Don't monitor devices

Definition at line 1004 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_QUALIFY_GAP   100

Definition at line 524 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_QUALIFY_PEERS   1

Definition at line 525 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_QUALIFYFREQ   60 * 1000

Qualification: How often to check for the host to be up

Definition at line 532 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_REALM   "asterisk"

Realm for HTTP digest authentication

Definition at line 998 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_REGEXTENONQUALIFY   FALSE

Definition at line 1007 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_REGISTRATION_TIMEOUT   20

Definition at line 501 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_RETRANS   1000

How frequently to retransmit Default: 2 * 500 ms in RFC 3261

Definition at line 535 of file chan_sip.c.

#define DEFAULT_RTPKEEPALIVE   0

Default RTPkeepalive setting

Definition at line 985 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_SDPOWNER   "root"

Default SDP username field in (o=) header unless re-defined in sip.conf

Definition at line 1013 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_SDPSESSION   "Asterisk PBX"

Default SDP session name, (s=) header unless re-defined in sip.conf

Definition at line 1012 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_SRVLOOKUP   TRUE

Recommended setting is ON

Definition at line 987 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_T1MIN   100

100 MS for minimal roundtrip time

Definition at line 1008 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_TIMER_T1   500

SIP timer T1 (according to RFC 3261)

Definition at line 537 of file chan_sip.c.

Referenced by reload_config().

#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.

Definition at line 990 of file chan_sip.c.

Referenced by reload_config().

#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.

Definition at line 989 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_TOS_TEXT   0

Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions.

Definition at line 992 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_TOS_VIDEO   0

Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions.

Definition at line 991 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_TRANS_TIMEOUT   -1

Use default SIP transaction timeout

Definition at line 541 of file chan_sip.c.

Referenced by __sip_autodestruct(), auto_congest(), cb_extensionstate(), check_auth(), check_pendings(), handle_incoming(), handle_invite_replaces(), handle_request_bye(), handle_request_cancel(), handle_request_info(), handle_request_invite(), handle_request_notify(), handle_request_options(), handle_request_register(), handle_response_invite(), handle_response_register(), receive_message(), sip_hangup(), sip_send_mwi_to_peer(), and transmit_fake_auth_response().

#define DEFAULT_USERAGENT   "Asterisk PBX"

Default Useragent: header unless re-defined in sip.conf

Definition at line 1011 of file chan_sip.c.

Referenced by reload_config().

#define DEFAULT_VMEXTEN   "asterisk"

Default voicemail extension

Definition at line 981 of file chan_sip.c.

Referenced by reload_config().

#define EXPIRY_GUARD_LIMIT   30

Below here, we use EXPIRY_GUARD_PCT instead of EXPIRY_GUARD_SECS

Definition at line 509 of file chan_sip.c.

Referenced by handle_response_register().

#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.

Definition at line 511 of file chan_sip.c.

Referenced by handle_response_register().

#define EXPIRY_GUARD_PCT   0.20

Percentage of expires timeout to use when below EXPIRY_GUARD_LIMIT

Definition at line 515 of file chan_sip.c.

Referenced by handle_response_register().

#define EXPIRY_GUARD_SECS   15

How long before expiry do we reregister

Definition at line 508 of file chan_sip.c.

Referenced by handle_response_register().

#define FALSE   0

Definition at line 477 of file chan_sip.c.

#define FINDALLDEVICES   (FINDUSERS | FINDPEERS)

Definition at line 487 of file chan_sip.c.

Referenced by check_peer_ok(), find_by_name(), find_peer(), sip_devicestate(), sip_peer_hold(), and update_call_counter().

#define FINDPEERS   (1 << 1)

Definition at line 486 of file chan_sip.c.

Referenced by _sip_qualify_peer(), _sip_show_peer(), check_peer_ok(), create_addr(), find_by_name(), find_peer(), function_sippeer(), register_verify(), sip_do_debug_peer(), sip_unregister(), st_get_mode(), st_get_refresher(), st_get_se(), and transmit_register().

#define FINDUSERS   (1 << 0)

Definition at line 485 of file chan_sip.c.

Referenced by check_peer_ok(), find_by_name(), find_peer(), and sip_show_user().

#define FORMAT   "%-15.15s %-15.15s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s\n"

Definition at line 16850 of file chan_sip.c.

#define FORMAT   "%-30.30s %-12.12s %-10.10s %-10.10s\n"

Definition at line 16850 of file chan_sip.c.

#define FORMAT   "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.6u %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.6u\n"

Definition at line 16850 of file chan_sip.c.

#define FORMAT   "%-30.30s %-6.6s %-12.12s %8d %-20.20s %-25.25s\n"

Definition at line 16850 of file chan_sip.c.

#define FORMAT   "%-40.40s %-20.20s %-16.16s\n"

Definition at line 16850 of file chan_sip.c.

#define FORMAT   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %s\n"

Definition at line 16850 of file chan_sip.c.

#define FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"

Definition at line 16850 of file chan_sip.c.

#define FORMAT   "%-30.30s %-6d %-9.9s %-6.6s\n"

Definition at line 16850 of file chan_sip.c.

#define FORMAT   "%-25.25s %-15.15s %-15.15s \n"

Definition at line 16850 of file chan_sip.c.

#define FORMAT2   "%-15.15s %-15.15s %-15.15s %-15.15s %-7.7s %-15.15s %-6.6s\n"

Definition at line 16849 of file chan_sip.c.

#define FORMAT2   "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s ( %%) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"

Definition at line 16849 of file chan_sip.c.

#define FORMAT2   "%-30.30s %-6.6s %-12.12s %8.8s %-20.20s %-25.25s\n"

Definition at line 16849 of file chan_sip.c.

#define FORMAT2   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %s\n"

Definition at line 16849 of file chan_sip.c.

#define FORMAT2   "%-30.30s %3.6s %9.9s %6.6s\n"

Definition at line 16849 of file chan_sip.c.

#define FORMAT2   "%-25.25s %-15.15s %-15.15s \n"

Definition at line 16849 of file chan_sip.c.

#define FORMAT3   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"

Definition at line 16848 of file chan_sip.c.

Referenced by sip_show_channels().

#define FORMAT4   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"

Definition at line 16847 of file chan_sip.c.

Referenced by show_channels_cb().

#define FROMDOMAIN_INVALID   "anonymous.invalid"

Definition at line 529 of file chan_sip.c.

Referenced by initreqprep().

#define INC_CALL_LIMIT   1

Definition at line 1166 of file chan_sip.c.

Referenced by handle_request_invite(), and update_call_counter().

#define INC_CALL_RINGING   3

Definition at line 1168 of file chan_sip.c.

Referenced by sip_call(), and update_call_counter().

#define INITIAL_CSEQ   101

Our initial sip sequence number

Definition at line 550 of file chan_sip.c.

Referenced by sip_alloc(), sip_register(), and transmit_response_using_temp().

#define IS_SIP_TECH

(

t

)

((t) == &sip_tech || (t) == &sip_tech_info)

Definition at line 2741 of file chan_sip.c.

Referenced by acf_channel_read(), func_header_read(), function_sipchaninfo_read(), handle_request_bye(), sip_dtmfmode(), and sip_hangup().

#define MAX_AUTHTRIES   3

Try authentication three times, then fail

Definition at line 543 of file chan_sip.c.

Referenced by handle_response(), handle_response_invite(), and handle_response_register().

#define MAX_HISTORY_ENTRIES   50

Max entires in the history list for a sip_pvt

Definition at line 548 of file chan_sip.c.

Referenced by append_history_va().

#define MAX_RETRANS   6

Try only 6 times for retransmissions, a total of 7 transmissions

Definition at line 536 of file chan_sip.c.

#define NO_RTP   0

Definition at line 576 of file chan_sip.c.

#define NOT_SUPPORTED   0

Definition at line 870 of file chan_sip.c.

#define PROVIS_KEEPALIVE_TIMEOUT   60000

How long to wait before retransmitting a provisional response (rfc 3261 13.3.1.1)

Definition at line 542 of file chan_sip.c.

Referenced by send_provisional_keepalive_full(), and update_provisional_keepalive().

#define REQ_OFFSET_TO_STR	(	req,
		offset		)	(ast_str_buffer((req)->data) + ((req)->offset))

Definition at line 1233 of file chan_sip.c.

Referenced by __get_header(), _sip_tcp_helper_thread(), find_sdp(), get_body(), get_destination(), get_msg_text(), get_sdp_iterate(), get_sdp_line(), handle_incoming(), handle_request_do(), handle_request_invite(), reqprep(), and send_response().

#define RTP   1

Definition at line 575 of file chan_sip.c.

#define SDP_MAX_RTPMAP_CODECS   32

Maximum number of codecs allowed in received SDP

Definition at line 555 of file chan_sip.c.

Referenced by process_sdp_a_audio(), process_sdp_a_text(), and process_sdp_a_video().

#define SIP_CALL_LIMIT   (1 << 7)

D: Call limit enforced for this call

Definition at line 1327 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), and update_call_counter().

#define SIP_DEFER_BYE_ON_TRANSFER   (1 << 10)

D: Do not hangup at first ast_hangup

Definition at line 1330 of file chan_sip.c.

Referenced by handle_invite_replaces(), handle_request_refer(), local_attended_transfer(), sip_hangup(), and sip_set_rtp_peer().

#define SIP_DIRECT_MEDIA   (1 << 20)

DP: allow peers to be reinvited to send media directly p2p

Definition at line 1355 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), reload_config(), sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), and sip_get_vrtp_peer().

#define SIP_DIRECT_MEDIA_NAT   (2 << 20)

DP: allow media reinvite when new peer is behind NAT

Definition at line 1356 of file chan_sip.c.

Referenced by handle_common_options(), sip_get_rtp_peer(), and sip_set_rtp_peer().

#define SIP_DTMF   (7 << 15)

DP: DTMF Support: five settings, uses three bits

Definition at line 1338 of file chan_sip.c.

Referenced by _sip_show_peer(), check_peer_ok(), create_addr_from_peer(), handle_common_options(), handle_request_invite(), process_sdp(), sip_alloc(), sip_dtmfmode(), sip_new(), sip_rtp_read(), sip_senddigit_begin(), sip_senddigit_end(), sip_show_channel(), sip_show_settings(), and transmit_info_with_digit().

#define SIP_DTMF_AUTO   (3 << 15)

DP: DTMF Support: AUTO switch between rfc2833 and in-band DTMF

Definition at line 1342 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), handle_common_options(), process_sdp(), sip_alloc(), and sip_new().

#define SIP_DTMF_INBAND   (1 << 15)

DP: DTMF Support: Inband audio, only for ULAW/ALAW - "inband"

Definition at line 1340 of file chan_sip.c.

Referenced by handle_common_options(), process_sdp(), sip_dtmfmode(), sip_new(), sip_rtp_read(), sip_senddigit_begin(), and sip_senddigit_end().

#define SIP_DTMF_INFO   (2 << 15)

DP: DTMF Support: SIP Info messages - "info"

Definition at line 1341 of file chan_sip.c.

Referenced by handle_common_options(), sip_dtmfmode(), sip_new(), and sip_senddigit_end().

#define SIP_DTMF_RFC2833   (0 << 15)

DP: DTMF Support: RTP DTMF - "rfc2833"

Definition at line 1339 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), handle_common_options(), handle_request_invite(), process_sdp(), reload_config(), sip_alloc(), sip_dtmfmode(), sip_rtp_read(), sip_senddigit_begin(), and sip_senddigit_end().

#define SIP_DTMF_SHORTINFO   (4 << 15)

DP: DTMF Support: SIP Info messages - "info" - short variant

Definition at line 1343 of file chan_sip.c.

Referenced by handle_common_options(), sip_dtmfmode(), sip_new(), sip_senddigit_end(), and transmit_info_with_digit().

#define SIP_FLAGS_TO_COPY

Value:

(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
    SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
    SIP_USEREQPHONE | SIP_INSECURE)

Flags to copy from peer/user to dialog.

Definition at line 1375 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), set_peer_defaults(), sip_alloc(), and sip_poke_peer().

#define SIP_G726_NONSTANDARD   (1 << 31)

DP: Use non-standard packing for G726-32 data

Definition at line 1372 of file chan_sip.c.

Referenced by add_codec_to_sdp(), handle_common_options(), and process_sdp_a_audio().

#define SIP_GOTREFER   (1 << 6)

D: Got a refer?

Definition at line 1326 of file chan_sip.c.

Referenced by handle_request_refer(), local_attended_transfer(), sip_set_rtp_peer(), and sip_set_udptl_peer().

#define SIP_INC_COUNT   (1 << 8)

D: Did this dialog increment the counter of in-use calls?

Definition at line 1328 of file chan_sip.c.

Referenced by __sip_destroy(), handle_request_cancel(), sip_hangup(), and update_call_counter().

#define SIP_INC_RINGING   (1 << 9)

D: Did this connection increment the counter of in-use calls?

Definition at line 1329 of file chan_sip.c.

Referenced by update_call_counter().

#define SIP_INSECURE   (3 << 23)

DP: three settings, uses two bits

Definition at line 1360 of file chan_sip.c.

Referenced by _sip_show_peer(), and handle_common_options().

#define SIP_INSECURE_INVITE   (1 << 24)

DP: don't require authentication for incoming INVITEs

Definition at line 1363 of file chan_sip.c.

Referenced by check_peer_ok(), and set_insecure_flags().

#define SIP_INSECURE_NONE   (0 << 23)

DP: secure mode

Definition at line 1361 of file chan_sip.c.

#define SIP_INSECURE_PORT   (1 << 23)

DP: don't require matching port for incoming requests

Definition at line 1362 of file chan_sip.c.

Referenced by find_peer(), get_insecure_variable_from_config(), peer_ipcmp_cb(), and set_insecure_flags().

#define SIP_MAX_HEADERS   64

Max amount of SIP headers to read

Definition at line 545 of file chan_sip.c.

Referenced by add_header(), and parse_request().

#define SIP_MAX_LINES   256

Max amount of lines in SIP attachment (like SDP)

Definition at line 546 of file chan_sip.c.

Referenced by parse_request().

#define SIP_MIN_PACKET   4096

Initialize size of memory to allocate for packets

Definition at line 547 of file chan_sip.c.

Referenced by _sip_tcp_helper_thread(), init_req(), init_resp(), and sipsock_read().

#define SIP_NAT   (3 << 18)

DP: four settings, uses two bits

Definition at line 1346 of file chan_sip.c.

Referenced by _sip_show_peer(), build_via(), copy_via_headers(), create_addr(), create_addr_from_peer(), display_nat_warning(), handle_common_options(), process_sdp(), register_verify(), sip_alloc(), sip_nat_mode(), sip_real_dst(), sip_show_channel(), sip_show_settings(), sip_show_users(), and transmit_response_using_temp().

#define SIP_NAT_ALWAYS   (3 << 18)

DP: NAT Both ROUTE and RFC3581

Definition at line 1350 of file chan_sip.c.

Referenced by copy_via_headers(), handle_common_options(), and reload_config().

#define SIP_NAT_NEVER   (0 << 18)

DP: No nat support

Definition at line 1347 of file chan_sip.c.

Referenced by handle_common_options().

#define SIP_NAT_RFC3581   (1 << 18)

DP: NAT RFC3581

Definition at line 1348 of file chan_sip.c.

Referenced by build_via(), copy_via_headers(), and handle_common_options().

#define SIP_NAT_ROUTE   (2 << 18)

DP: NAT Only ROUTE

Definition at line 1349 of file chan_sip.c.

Referenced by _sip_show_peers(), build_peer(), check_peer_ok(), check_user_full(), create_addr(), create_addr_from_peer(), handle_common_options(), parse_register_contact(), send_request(), set_address_from_contact(), sip_alloc(), sip_nat_mode(), sip_real_dst(), and transmit_response_using_temp().

#define SIP_NEEDREINVITE   (1 << 4)

D: Do we need to send another reinvite?

Definition at line 1324 of file chan_sip.c.

Referenced by check_pendings(), interpret_t38_parameters(), sip_hangup(), sip_reinvite_retry(), sip_sendhtml(), sip_set_rtp_peer(), and sip_set_udptl_peer().

#define SIP_OPT_100REL   (1 << 1)

Definition at line 874 of file chan_sip.c.

#define SIP_OPT_EARLY_SESSION   (1 << 3)

Definition at line 876 of file chan_sip.c.

#define SIP_OPT_EVENTLIST   (1 << 11)

Definition at line 884 of file chan_sip.c.

#define SIP_OPT_FROMCHANGE   (1 << 17)

Definition at line 890 of file chan_sip.c.

#define SIP_OPT_GRUU   (1 << 12)

Definition at line 885 of file chan_sip.c.

#define SIP_OPT_HISTINFO   (1 << 15)

Definition at line 888 of file chan_sip.c.

#define SIP_OPT_JOIN   (1 << 4)

Definition at line 877 of file chan_sip.c.

#define SIP_OPT_NOREFERSUB   (1 << 14)

Definition at line 887 of file chan_sip.c.

#define SIP_OPT_OUTBOUND   (1 << 20)

Definition at line 893 of file chan_sip.c.

#define SIP_OPT_PATH   (1 << 5)

Definition at line 878 of file chan_sip.c.

#define SIP_OPT_PRECONDITION   (1 << 7)

Definition at line 880 of file chan_sip.c.

#define SIP_OPT_PREF   (1 << 6)

Definition at line 879 of file chan_sip.c.

#define SIP_OPT_PRIVACY   (1 << 8)

Definition at line 881 of file chan_sip.c.

#define SIP_OPT_RECLISTINV   (1 << 18)

Definition at line 891 of file chan_sip.c.

#define SIP_OPT_RECLISTSUB   (1 << 19)

Definition at line 892 of file chan_sip.c.

#define SIP_OPT_REPLACES   (1 << 0)

Definition at line 873 of file chan_sip.c.

Referenced by handle_request_invite().

#define SIP_OPT_RESPRIORITY   (1 << 16)

Definition at line 889 of file chan_sip.c.

#define SIP_OPT_SDP_ANAT   (1 << 9)

Definition at line 882 of file chan_sip.c.

#define SIP_OPT_SEC_AGREE   (1 << 10)

Definition at line 883 of file chan_sip.c.

#define SIP_OPT_TARGET_DIALOG   (1 << 13)

Definition at line 886 of file chan_sip.c.

#define SIP_OPT_TIMER   (1 << 2)

Definition at line 875 of file chan_sip.c.

Referenced by handle_request_invite().

#define SIP_OPT_UNKNOWN   (1 << 21)

Definition at line 894 of file chan_sip.c.

Referenced by parse_sip_options().

#define SIP_OUTGOING   (1 << 0)

D: Direction of the last transaction in this dialog

Definition at line 1321 of file chan_sip.c.

Referenced by __sip_subscribe_mwi_do(), handle_request_bye(), handle_request_invite(), handle_request_refer(), handle_response(), handle_response_invite(), manager_sipnotify(), reqprep(), respprep(), sip_call(), sip_cli_notify(), sip_hangup(), sip_indicate(), sip_poke_peer(), sip_send_mwi_to_peer(), sip_show_channel(), sip_write(), transmit_refer(), transmit_register(), transmit_reinvite_with_sdp(), and transmit_response_with_sdp().

#define SIP_PAGE2_ALLOWOVERLAP   (1 << 17)

DP: Allow overlap dialing ?

Definition at line 1396 of file chan_sip.c.

Referenced by _sip_show_peer(), get_destination(), handle_common_options(), handle_request_invite(), reload_config(), and sip_show_settings().

#define SIP_PAGE2_ALLOWSUBSCRIBE   (1 << 16)

GP: Allow subscriptions from this peer?

Definition at line 1395 of file chan_sip.c.

Referenced by _sip_show_peer(), build_peer(), handle_common_options(), handle_request_subscribe(), reload_config(), and sip_show_settings().

#define SIP_PAGE2_BUGGY_MWI   (1 << 26)

DP: Buggy CISCO MWI fix

Definition at line 1411 of file chan_sip.c.

Referenced by handle_common_options(), and transmit_notify_with_mwi().

#define SIP_PAGE2_CALL_ONHOLD   (3 << 23)

D: Call hold states:

Definition at line 1405 of file chan_sip.c.

Referenced by __sip_destroy(), add_sdp(), change_hold_state(), check_rtp_timeout(), handle_request_cancel(), handle_request_invite(), process_sdp(), show_channels_cb(), sip_hangup(), and update_call_counter().

#define SIP_PAGE2_CALL_ONHOLD_ACTIVE   (1 << 23)

D: Active hold

Definition at line 1406 of file chan_sip.c.

Referenced by change_hold_state().

#define SIP_PAGE2_CALL_ONHOLD_INACTIVE   (3 << 23)

D: Inactive hold

Definition at line 1408 of file chan_sip.c.

Referenced by add_sdp(), and change_hold_state().

#define SIP_PAGE2_CALL_ONHOLD_ONEDIR   (2 << 23)

D: One directional hold

Definition at line 1407 of file chan_sip.c.

Referenced by add_sdp(), and change_hold_state().

#define SIP_PAGE2_DIALOG_ESTABLISHED   (1 << 27)

29: Has a dialog been established?

Definition at line 1412 of file chan_sip.c.

Referenced by find_call_cb(), handle_request_bye(), handle_request_invite(), handle_request_subscribe(), handle_response(), handle_response_invite(), and sip_answer().

#define SIP_PAGE2_FAX_DETECT   (3 << 28)

DP: Fax Detection support

Definition at line 1413 of file chan_sip.c.

Referenced by handle_common_options(), reload_config(), sip_new(), and sip_rtp_read().

#define SIP_PAGE2_FAX_DETECT_BOTH   (3 << 28)

DP: Fax Detection support - detect both

Definition at line 1416 of file chan_sip.c.

Referenced by handle_common_options().

#define SIP_PAGE2_FAX_DETECT_CNG   (1 << 28)

DP: Fax Detection support - detect CNG in audio

Definition at line 1414 of file chan_sip.c.

Referenced by handle_common_options(), and sip_read().

#define SIP_PAGE2_FAX_DETECT_T38   (2 << 28)

DP: Fax Detection support - detect T.38 reinvite from peer

Definition at line 1415 of file chan_sip.c.

Referenced by handle_common_options(), process_sdp(), and sip_call().

#define SIP_PAGE2_FLAGS_TO_COPY

Value:

(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
   SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
   SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
   SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | \
   SIP_PAGE2_HAVEPEERCONTEXT | SIP_PAGE2_FORWARD_LOOP_DETECTED)

Definition at line 1421 of file chan_sip.c.

Referenced by check_peer_ok(), create_addr_from_peer(), set_peer_defaults(), sip_alloc(), and sip_poke_peer().

#define SIP_PAGE2_FORWARD_LOOP_DETECTED   (1 << 8)

31: Do call forward when receiving 482 Loop Detected

Definition at line 1389 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_response(), handle_t38_options(), reload_config(), and sip_show_settings().

#define SIP_PAGE2_HAVEPEERCONTEXT   (1 << 3)

Definition at line 1387 of file chan_sip.c.

Referenced by build_peer(), and get_destination().

#define SIP_PAGE2_IGNORESDPVERSION   (1 << 19)

GDP: Ignore the SDP session version number we receive and treat all sessions as new

Definition at line 1398 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), process_sdp_o(), reload_config(), and sip_show_settings().

#define SIP_PAGE2_REGISTERTRYING   (1 << 29)

DP: Send 100 Trying on REGISTER attempts

Definition at line 1417 of file chan_sip.c.

Referenced by _sip_show_peer(), build_peer(), and register_verify().

#define SIP_PAGE2_RFC2833_COMPENSATE   (1 << 25)

DP: Compensate for buggy RFC2833 implementations

Definition at line 1410 of file chan_sip.c.

Referenced by create_addr_from_peer(), handle_common_options(), handle_request_invite(), process_sdp(), sip_alloc(), and sip_show_settings().

#define SIP_PAGE2_RPORT_PRESENT   (1 << 10)

Was rport received in the Via header?

Definition at line 1392 of file chan_sip.c.

Referenced by check_user_full(), and check_via().

#define SIP_PAGE2_RTAUTOCLEAR   (1 << 2)

GP: Should we clean memory from peers after expiry?

Definition at line 1386 of file chan_sip.c.

Referenced by expire_register(), realtime_peer(), reload_config(), and sip_show_settings().

#define SIP_PAGE2_RTCACHEFRIENDS   (1 << 0)

GP: Should we keep RT objects in memory for extended time?

Definition at line 1385 of file chan_sip.c.

Referenced by build_peer(), parse_register_contact(), realtime_peer(), reload_config(), sip_destroy_peer(), sip_prune_realtime(), sip_show_settings(), and update_peer().

#define SIP_PAGE2_STATECHANGEQUEUE   (1 << 9)

D: Unsent state pending change exists

Definition at line 1390 of file chan_sip.c.

Referenced by cb_extensionstate(), handle_response(), and handle_response_notify().

#define SIP_PAGE2_SUBSCRIBEMWIONLY   (1 << 18)

GP: Only issue MWI notification if subscribed to

Definition at line 1397 of file chan_sip.c.

Referenced by build_peer(), handle_request_subscribe(), register_verify(), and sip_send_mwi_to_peer().

#define SIP_PAGE2_T38SUPPORT   (3 << 20)

GDP: T.38 Fax Support

Definition at line 1400 of file chan_sip.c.

Referenced by _sip_show_peer(), check_peer_ok(), create_addr_from_peer(), handle_request_invite(), handle_t38_options(), interpret_t38_parameters(), set_t38_capabilities(), sip_alloc(), sip_queryoption(), and sip_show_settings().

#define SIP_PAGE2_T38SUPPORT_UDPTL   (1 << 20)

GDP: T.38 Fax Support (no error correction)

Definition at line 1401 of file chan_sip.c.

Referenced by handle_t38_options(), and set_t38_capabilities().

#define SIP_PAGE2_T38SUPPORT_UDPTL_FEC   (2 << 20)

GDP: T.38 Fax Support (FEC error correction)

Definition at line 1402 of file chan_sip.c.

Referenced by handle_t38_options(), and set_t38_capabilities().

#define SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY   (3 << 20)

GDP: T.38 Fax Support (redundancy error correction)

Definition at line 1403 of file chan_sip.c.

Referenced by handle_t38_options(), and set_t38_capabilities().

#define SIP_PAGE2_TEXTSUPPORT   (1 << 15)

GDP: Global text enable

Definition at line 1394 of file chan_sip.c.

Referenced by _sip_show_peer(), _sip_show_peers(), check_peer_ok(), create_addr_from_peer(), handle_common_options(), reload_config(), sip_alloc(), and sip_show_settings().

#define SIP_PAGE2_UDPTL_DESTINATION   (1 << 30)

DP: Use source IP of RTP as destination if NAT is enabled

Definition at line 1418 of file chan_sip.c.

Referenced by handle_t38_options(), and process_sdp().

#define SIP_PAGE2_VIDEOSUPPORT   (1 << 14)

DP: Video supported if offered?

Definition at line 1393 of file chan_sip.c.

Referenced by _sip_show_peer(), _sip_show_peers(), check_peer_ok(), create_addr_from_peer(), handle_common_options(), reload_config(), sip_alloc(), sip_new(), and sip_show_settings().

#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS   (1 << 31)

DP: Always set up video, even if endpoints don't support it

Definition at line 1419 of file chan_sip.c.

Referenced by _sip_show_peer(), check_peer_ok(), create_addr_from_peer(), handle_common_options(), reload_config(), and sip_alloc().

#define SIP_PENDINGBYE   (1 << 5)

D: Need to send bye after we ack?

Definition at line 1325 of file chan_sip.c.

Referenced by check_pendings(), handle_response_invite(), interpret_t38_parameters(), register_verify(), sip_hangup(), sip_sendhtml(), sip_set_rtp_peer(), and sip_set_udptl_peer().

#define SIP_PROG_INBAND   (3 << 25)

DP: three settings, uses two bits

Definition at line 1366 of file chan_sip.c.

Referenced by handle_common_options(), sip_indicate(), and sip_show_settings().

#define SIP_PROG_INBAND_NEVER   (0 << 25)

Definition at line 1367 of file chan_sip.c.

Referenced by sip_indicate(), and sip_show_settings().

#define SIP_PROG_INBAND_NO   (1 << 25)

Definition at line 1368 of file chan_sip.c.

Referenced by handle_common_options(), and sip_show_settings().

#define SIP_PROG_INBAND_YES   (2 << 25)

Definition at line 1369 of file chan_sip.c.

Referenced by handle_common_options(), and sip_indicate().

#define SIP_PROGRESS_SENT   (1 << 3)

D: Have sent 183 message progress

Definition at line 1323 of file chan_sip.c.

Referenced by sip_indicate(), and sip_write().

#define SIP_PROMISCREDIR   (1 << 11)

DP: Promiscuous redirection

Definition at line 1332 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), parse_moved_contact(), sip_show_channel(), and sip_show_settings().

#define sip_pvt_lock

(

x

)

ao2_lock(x)

Definition at line 1747 of file chan_sip.c.

Referenced by __sip_ack(), __sip_autodestruct(), auto_congest(), cb_extensionstate(), check_rtp_timeout(), complete_sipch(), find_call(), get_sip_pvt_byid_locked(), handle_invite_replaces(), handle_request_bye(), handle_request_invite(), proc_session_timer(), retrans_pkt(), scheduler_process_request_queue(), sip_answer(), sip_dtmfmode(), sip_fixup(), sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), sip_get_vrtp_peer(), sip_hangup(), sip_indicate(), sip_new(), sip_park(), sip_queryoption(), sip_read(), sip_reg_timeout(), sip_reinvite_retry(), sip_request_call(), sip_senddigit_begin(), sip_senddigit_end(), sip_set_rtp_peer(), sip_set_udptl_peer(), sip_show_channel(), sip_show_history(), sip_t38_abort(), sip_transfer(), sip_write(), and update_call_counter().

#define sip_pvt_trylock

(

x

)

ao2_trylock(x)

Definition at line 1748 of file chan_sip.c.

Referenced by dialog_needdestroy(), find_call(), and sip_hangup().

#define sip_pvt_unlock

(

x

)

ao2_unlock(x)

Definition at line 1749 of file chan_sip.c.

Referenced by __sip_ack(), __sip_autodestruct(), auto_congest(), cb_extensionstate(), check_rtp_timeout(), complete_sipch(), dialog_needdestroy(), get_sip_pvt_byid_locked(), handle_invite_replaces(), handle_request_bye(), handle_request_do(), handle_request_invite(), local_attended_transfer(), proc_session_timer(), retrans_pkt(), scheduler_process_request_queue(), sip_answer(), sip_dtmfmode(), sip_fixup(), sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), sip_get_vrtp_peer(), sip_hangup(), sip_indicate(), sip_new(), sip_park(), sip_queryoption(), sip_read(), sip_reg_timeout(), sip_reinvite_retry(), sip_request_call(), sip_senddigit_begin(), sip_senddigit_end(), sip_set_rtp_peer(), sip_set_udptl_peer(), sip_show_channel(), sip_show_history(), sip_t38_abort(), sip_transfer(), sip_write(), and update_call_counter().

#define SIP_REINVITE   (7 << 20)

DP: four settings, uses three bits

Definition at line 1353 of file chan_sip.c.

Referenced by handle_common_options().

#define SIP_REINVITE_NONE   (0 << 20)

DP: no reinvite allowed

Definition at line 1354 of file chan_sip.c.

#define SIP_REINVITE_UPDATE   (4 << 20)

DP: use UPDATE (RFC3311) when reinviting this peer

Definition at line 1357 of file chan_sip.c.

Referenced by handle_common_options(), and transmit_reinvite_with_sdp().

#define SIP_RESERVED   ";/?:@&=+$,# "

Reserved characters in the username part of the URI

Definition at line 493 of file chan_sip.c.

#define SIP_RINGING   (1 << 2)

D: Have sent 180 ringing

Definition at line 1322 of file chan_sip.c.

Referenced by sip_indicate().

#define SIP_SENDRPID   (1 << 29)

DP: Remote Party-ID Support

Definition at line 1371 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), and initreqprep().

#define SIP_TRANS_TIMEOUT   64 * DEFAULT_TIMER_T1

SIP request timeout (rfc 3261) 64*T1

Todo:

Use known T1 for timeout (peerpoke)

Definition at line 538 of file chan_sip.c.

Referenced by manager_sipnotify(), sip_cli_notify(), and sip_sipredirect().

#define SIP_TRUSTRPID   (1 << 12)

DP: Trust RPID headers?

Definition at line 1333 of file chan_sip.c.

Referenced by _sip_show_peer(), handle_common_options(), and replace_cid().

#define SIP_USECLIENTCODE   (1 << 14)

DP: Trust X-ClientCode info message

Definition at line 1335 of file chan_sip.c.

Referenced by handle_common_options(), handle_request_info(), and sip_show_settings().

#define SIP_USEREQPHONE   (1 << 13)

DP: Add user=phone to numeric URI. Default off

Definition at line 1334 of file chan_sip.c.

Referenced by _sip_show_peer(), build_peer(), initreqprep(), reload_config(), and sip_show_settings().

#define SIPBUFSIZE   512

Buffer size for many operations

Definition at line 489 of file chan_sip.c.

Referenced by add_route(), add_sdp(), build_contact(), extract_uri(), handle_request_refer(), initreqprep(), parse_moved_contact(), parse_ok_contact(), parse_register_contact(), process_sdp(), respprep(), show_channels_cb(), sip_call(), sip_new(), sip_show_channel(), sip_show_settings(), transmit_invite(), and transmit_notify_with_sipfrag().

#define STANDARD_SIP_PORT   5060

Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS.

Definition at line 958 of file chan_sip.c.

Referenced by __set_address_from_contact(), build_peer(), check_via(), create_addr(), manager_show_registry(), parse_register_contact(), process_via(), proxy_allocate(), reload_config(), set_destination(), set_peer_defaults(), sip_parse_host(), sip_register(), sip_show_mwi(), sip_show_registry(), sip_standard_port(), and transmit_register().

#define STANDARD_TLS_PORT   5061

Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS.

Definition at line 960 of file chan_sip.c.

Referenced by __set_address_from_contact(), build_peer(), create_addr(), parse_register_contact(), reload_config(), set_destination(), sip_parse_host(), sip_register(), and sip_standard_port().

#define SUPPORTED   1

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

Definition at line 869 of file chan_sip.c.

#define TRUE   1

Definition at line 481 of file chan_sip.c.

#define UNLINK	(	element,
		head,
		prev		)

some list management macros.

Definition at line 2240 of file chan_sip.c.

Referenced by __sip_ack(), handle_request_cancel(), and retrans_pkt().

#define XMIT_ERROR   -2

Definition at line 491 of file chan_sip.c.

Referenced by __sip_reliable_xmit(), __sip_xmit(), handle_response_invite(), retrans_pkt(), sip_call(), sip_poke_peer(), and sip_tcptls_write().

---

Enumeration Type Documentation

enum can_create_dialog

States whether a SIP message can create a dialog in Asterisk.

Enumerator:

CAN_NOT_CREATE_DIALOG
CAN_CREATE_DIALOG
CAN_CREATE_DIALOG_UNSUPPORTED_METHOD

Definition at line 785 of file chan_sip.c.

                       {
   CAN_NOT_CREATE_DIALOG,
   CAN_CREATE_DIALOG,
   CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
};

enum check_auth_result

Authentication result from check_auth* functions.

Enumerator:

AUTH_DONT_KNOW	no result, need to check further
AUTH_SUCCESSFUL
AUTH_CHALLENGE_SENT
AUTH_SECRET_FAILED
AUTH_USERNAME_MISMATCH
AUTH_NOT_FOUND	returned by register_verify
AUTH_FAKE_AUTH
AUTH_UNKNOWN_DOMAIN
AUTH_PEER_NOT_DYNAMIC
AUTH_ACL_FAILED
AUTH_BAD_TRANSPORT

Definition at line 685 of file chan_sip.c.

                       {
   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,
};

enum domain_mode

Modes for SIP domain handling in the PBX.

Enumerator:

SIP_DOMAIN_AUTO	This domain is auto-configured
SIP_DOMAIN_CONFIG	This domain is from configuration

Definition at line 1275 of file chan_sip.c.

                 {
   SIP_DOMAIN_AUTO,     /*!< This domain is auto-configured */
   SIP_DOMAIN_CONFIG,      /*!< This domain is from configuration */
};

enum invitestates

States for the INVITE transaction, not the dialog.

Note:

this is for the INVITE that sets up the dialog

Enumerator:

INV_NONE	No state at all, maybe not an INVITE dialog
INV_CALLING	Invite sent, no answer
INV_PROCEEDING	We got/sent 1xx message
INV_EARLY_MEDIA	We got 18x message with to-tag back
INV_COMPLETED	Got final response with error. Wait for ACK, then CONFIRMED
INV_CONFIRMED	Confirmed response - we've got an ack (Incoming calls only)
INV_TERMINATED	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	Transaction cancelled by client or server in non-terminated state

Definition at line 597 of file chan_sip.c.

                  {
   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 */
};

enum media_type

Enumerator:

SDP_AUDIO
SDP_VIDEO
SDP_IMAGE
SDP_TEXT

Definition at line 8513 of file chan_sip.c.

                {
   SDP_AUDIO,
   SDP_VIDEO,
   SDP_IMAGE,
   SDP_TEXT,
};

enum notifycid_setting

Settings for the 'notifycid' option, see sip.conf.sample for details.

Enumerator:

DISABLED
ENABLED
IGNORE_CONTEXT

Definition at line 822 of file chan_sip.c.

                       {
   DISABLED       = 0,
   ENABLED        = 1,
   IGNORE_CONTEXT = 2,
};

enum parse_register_result

Results from the parse_register() function.

Enumerator:

PARSE_REGISTER_DENIED
PARSE_REGISTER_FAILED
PARSE_REGISTER_UPDATE
PARSE_REGISTER_QUERY

Definition at line 636 of file chan_sip.c.

                           {
   PARSE_REGISTER_DENIED,
   PARSE_REGISTER_FAILED,
   PARSE_REGISTER_UPDATE,
   PARSE_REGISTER_QUERY,
};

enum peer_unlink_flag_t

Enumerator:

SIP_PEERS_MARKED
SIP_PEERS_ALL

Definition at line 3319 of file chan_sip.c.

             {
   SIP_PEERS_MARKED,
   SIP_PEERS_ALL,
} peer_unlink_flag_t;

enum referstatus

Parameters to know status of transfer.

Enumerator:

REFER_IDLE	No REFER is in progress
REFER_SENT	Sent REFER to transferee
REFER_RECEIVED	Received REFER from transferrer
REFER_CONFIRMED	Refer confirmed with a 100 TRYING (unused)
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

Definition at line 1466 of file chan_sip.c.

                 {
   REFER_IDLE,                    /*!< No REFER is in progress */
   REFER_SENT,                    /*!< Sent REFER to transferee */
   REFER_RECEIVED,                /*!< Received REFER from transferrer */
   REFER_CONFIRMED,               /*!< Refer confirmed with a 100 TRYING (unused) */
   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 */
};

enum sip_auth_type

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.

Enumerator:

PROXY_AUTH
WWW_AUTH

Definition at line 679 of file chan_sip.c.

                   {
   PROXY_AUTH = 407,
   WWW_AUTH = 401,
};

enum sip_debug_e

debugging state We store separately the debugging requests from the config file and requests from the CLI. Debugging is enabled if either is set (which means that if sipdebug is set in the config file, we can only turn it off by reloading the config).

Enumerator:

sip_debug_none
sip_debug_config
sip_debug_console

Definition at line 1436 of file chan_sip.c.

                 {
   sip_debug_none = 0,
   sip_debug_config = 1,
   sip_debug_console = 2,
};

enum sip_peer_type

Enumerator:

SIP_TYPE_PEER
SIP_TYPE_USER

Definition at line 1830 of file chan_sip.c.

                   {
   SIP_TYPE_PEER = (1 << 0),
   SIP_TYPE_USER = (1 << 1),
};

enum sip_result

The result of a lot of functions.

Enumerator:

AST_SUCCESS	FALSE means success, funny enough
AST_FAILURE	Failure code

Definition at line 589 of file chan_sip.c.

                {
   AST_SUCCESS = 0,     /*!< FALSE means success, funny enough */
   AST_FAILURE = -1,    /*!< Failure code */
};

enum sip_tcptls_alert

Enumerator:

TCPTLS_ALERT_DATA	There is new data to be sent out.
TCPTLS_ALERT_STOP	A request to stop the tcp_handler thread.

Definition at line 1976 of file chan_sip.c.

                      {
   /*! \brief There is new data to be sent out */
   TCPTLS_ALERT_DATA,
   /*! \brief A request to stop the tcp_handler thread */
   TCPTLS_ALERT_STOP,
};

enum sip_transport

Define some implemented SIP transports.

Note:

Asterisk does not support SCTP or UDP/DTLS

Enumerator:

SIP_TRANSPORT_UDP	Unreliable transport for SIP, needs retransmissions
SIP_TRANSPORT_TCP	Reliable, but unsecure
SIP_TRANSPORT_TLS	TCP/TLS - reliable and secure transport for signalling

Definition at line 754 of file chan_sip.c.

                   {
   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 */
};

enum sipmethod

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.

Enumerator:

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

Definition at line 801 of file chan_sip.c.

               {
   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 */
};

enum sipregistrystate

States for outbound registrations (with register= lines in sip.conf.

Enumerator:

REG_STATE_UNREGISTERED	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

Definition at line 702 of file chan_sip.c.

                      {
   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 */
};

enum st_mode

Modes in which Asterisk can be configured to run SIP Session-Timers.

Enumerator:

SESSION_TIMER_MODE_INVALID	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

Definition at line 737 of file chan_sip.c.

             {
        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 */
};

enum st_refresher

The entity playing the refresher role for Session-Timers.

Enumerator:

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

Definition at line 745 of file chan_sip.c.

                  {
        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 */
};

enum subscriptiontype

Type of subscription, based on the packages we do support, see subscription_types.

Enumerator:

NONE
XPIDF_XML
DIALOG_INFO_XML
CPIM_PIDF_XML
PIDF_XML
MWI_NOTIFICATION

Definition at line 644 of file chan_sip.c.

                      { 
   NONE = 0,
   XPIDF_XML,
   DIALOG_INFO_XML,
   CPIM_PIDF_XML,
   PIDF_XML,
   MWI_NOTIFICATION
};

enum t38_action_flag

Enumerator:

SDP_T38_NONE	Do not modify T38 information at all
SDP_T38_INITIATE	Remote side has requested T38 with us
SDP_T38_ACCEPT	Remote side accepted our T38 request

Definition at line 2247 of file chan_sip.c.

                     {
   SDP_T38_NONE = 0, /*!< Do not modify T38 information at all */
   SDP_T38_INITIATE, /*!< Remote side has requested T38 with us */
   SDP_T38_ACCEPT,   /*!< Remote side accepted our T38 request */
};

enum t38state

T38 States for a call.

Enumerator:

T38_DISABLED	Not enabled
T38_LOCAL_REINVITE	Offered from local - REINVITE
T38_PEER_REINVITE	Offered from peer - REINVITE
T38_ENABLED	Negotiated (enabled)

Definition at line 1451 of file chan_sip.c.

              {
   T38_DISABLED = 0,                /*!< Not enabled */
   T38_LOCAL_REINVITE,              /*!< Offered from local - REINVITE */
   T38_PEER_REINVITE,               /*!< Offered from peer - REINVITE */
   T38_ENABLED                      /*!< Negotiated (enabled) */
};

enum transfermodes

Authorization scheme for call transfers.

Note:

Not a bitfield flag, since there are plans for other modes, like "only allow transfers for authenticated devices"

Enumerator:

TRANSFER_OPENFORALL	Allow all SIP transfers
TRANSFER_CLOSED	Allow no SIP transfers

Definition at line 582 of file chan_sip.c.

                   {
   TRANSFER_OPENFORALL,            /*!< Allow all SIP transfers */
   TRANSFER_CLOSED,                /*!< Allow no SIP transfers */
};

enum xmittype

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.

Enumerator:

XMIT_CRITICAL	Transmit critical SIP message reliably, with re-transmits. If it fails, it's critical and will cause a teardown of the session
XMIT_RELIABLE	Transmit SIP message reliably, with re-transmits
XMIT_UNRELIABLE	Transmit SIP message without bothering with re-transmits

Definition at line 628 of file chan_sip.c.

              {
   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 */
};

---

Function Documentation

static const char * __get_header	(	const struct sip_request *	req,
		const char *	name,
		int *	start
	)			[static]

Definition at line 7224 of file chan_sip.c.

References ast_skip_blanks(), find_alias(), sip_request::header, sip_request::headers, pass, sip_settings::pedanticsipchecking, REQ_OFFSET_TO_STR, and sip_cfg.

{
   int pass;

   /*
    * Technically you can place arbitrary whitespace both before and after the ':' in
    * a header, although RFC3261 clearly says you shouldn't before, and place just
    * one afterwards.  If you shouldn't do it, what absolute idiot decided it was 
    * a good idea to say you can do it, and if you can do it, why in the hell would.
    * you say you shouldn't.
    * Anyways, pedanticsipchecking controls whether we allow spaces before ':',
    * and we always allow spaces after that for compatibility.
    */
   for (pass = 0; name && pass < 2;pass++) {
      int x, len = strlen(name);
      for (x = *start; x < req->headers; x++) {
         char *header = REQ_OFFSET_TO_STR(req, header[x]);
         if (!strncasecmp(header, name, len)) {
            char *r = header + len; /* skip name */
            if (sip_cfg.pedanticsipchecking)
               r = ast_skip_blanks(r);

            if (*r == ':') {
               *start = x+1;
               return ast_skip_blanks(r+1);
            }
         }
      }
      if (pass == 0) /* Try aliases */
         name = find_alias(name, NULL);
   }

   /* Don't return NULL, so get_header is always a valid pointer */
   return "";
}

static void __init_check_auth_buf

(

void

)

[static]

Definition at line 13178 of file chan_sip.c.

{

static void __init_ts_temp_pvt

(

void

)

[static]

A per-thread temporary pvt structure.

Definition at line 2156 of file chan_sip.c.

: The address we bind to */

static void __reg_module

(

void

)

[static]

Definition at line 26863 of file chan_sip.c.

static int __set_address_from_contact	(	const char *	fullcontact,
		struct sockaddr_in *	sin,
		int	tcp
	)			[static]

Definition at line 12759 of file chan_sip.c.

References ast_copy_string(), ast_gethostbyname(), ast_log(), get_transport_str2enum(), hp, LOG_WARNING, parse_uri(), port_str2int(), SIP_TRANSPORT_TLS, STANDARD_SIP_PORT, and STANDARD_TLS_PORT.

Referenced by build_peer(), and set_address_from_contact().

{
   struct hostent *hp;
   struct ast_hostent ahp;
   int port = STANDARD_SIP_PORT;
   char *host, *pt, *transport;
   char contact_buf[256];
   char *contact;

   /* Work on a copy */
   ast_copy_string(contact_buf, fullcontact, sizeof(contact_buf));
   contact = contact_buf;

   /* 
    * We have only the part in <brackets> here so we just need to parse a SIP URI.
    *
    * Note: The outbound proxy could be using UDP between the proxy and Asterisk.
    * We still need to be able to send to the remote agent through the proxy.
    */

   if (parse_uri(contact, "sip:,sips:", &contact, NULL, &host, &pt, NULL, &transport)) {
      ast_log(LOG_WARNING, "Invalid contact uri %s (missing sip: or sips:), attempting to use anyway\n", fullcontact);
   }

   /* set port */
   if (((get_transport_str2enum(transport) == SIP_TRANSPORT_TLS)) || !(strncasecmp(fullcontact, "sips", 4))) {
      port = port_str2int(pt, STANDARD_TLS_PORT);
   } else {
      port = port_str2int(pt, STANDARD_SIP_PORT);
   }


   /* XXX This could block for a long time XXX */
   /* We should only do this if it's a name, not an IP */
   /* \todo - if there's no PORT number in contact - we are required to check NAPTR/SRV records
      to find transport, port address and hostname. If there's a port number, we have to
      assume that the domain part is a host name and only look for an A/AAAA record in DNS.
   */
   hp = ast_gethostbyname(host, &ahp);
   if (!hp)  {
      ast_log(LOG_WARNING, "Invalid host name in Contact: (can't resolve in DNS) : '%s'\n", host);
      return -1;
   }
   sin->sin_family = AF_INET;
   memcpy(&sin->sin_addr, hp->h_addr, sizeof(sin->sin_addr));
   sin->sin_port = htons(port);

   return 0;
}

static int __sip_ack	(	struct sip_pvt *	p,
		int	seqno,
		int	resp,
		int	sipmethod
	)			[static]

Acknowledges receipt of a packet and stops retransmission called with p locked.

Definition at line 4312 of file chan_sip.c.

References ast_debug, ast_free, ast_sched_del(), sip_pvt::callid, sip_pkt::data, dialog_unref(), FALSE, sip_proxy::force, sip_pkt::is_resp, sip_pkt::method, msg, sip_pkt::next, sip_pvt::outboundproxy, sip_pkt::owner, sip_pvt::packets, sip_pvt::pendinginvite, ref_proxy(), sip_pkt::retransid, sip_pkt::seqno, sip_pvt_lock, sip_pvt_unlock, TRUE, and UNLINK.

Referenced by __sip_pretend_ack(), handle_incoming(), handle_request_invite(), and handle_response().

{
   struct sip_pkt *cur, *prev = NULL;
   const char *msg = "Not Found";   /* used only for debugging */
   int res = FALSE;

   /* If we have an outbound proxy for this dialog, then delete it now since
     the rest of the requests in this dialog needs to follow the routing.
     If obforcing is set, we will keep the outbound proxy during the whole
     dialog, regardless of what the SIP rfc says
   */
   if (p->outboundproxy && !p->outboundproxy->force){
      ref_proxy(p, NULL);
   }

   for (cur = p->packets; cur; prev = cur, cur = cur->next) {
      if (cur->seqno != seqno || cur->is_resp != resp)
         continue;
      if (cur->is_resp || cur->method == sipmethod) {
         res = TRUE;
         msg = "Found";
         if (!resp && (seqno == p->pendinginvite)) {
            ast_debug(1, "Acked pending invite %d\n", p->pendinginvite);
            p->pendinginvite = 0;
         }
         if (cur->retransid > -1) {
            if (sipdebug)
               ast_debug(4, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
         }
         /* This odd section is designed to thwart a 
          * race condition in the packet scheduler. There are
          * two conditions under which deleting the packet from the
          * scheduler can fail.
          *
          * 1. The packet has been removed from the scheduler because retransmission
          * is being attempted. The problem is that if the packet is currently attempting
          * retransmission and we are at this point in the code, then that MUST mean
          * that retrans_pkt is waiting on p's lock. Therefore we will relinquish the
          * lock temporarily to allow retransmission.
          *
          * 2. The packet has reached its maximum number of retransmissions and has
          * been permanently removed from the packet scheduler. If this is the case, then
          * the packet's retransid will be set to -1. The atomicity of the setting and checking
          * of the retransid to -1 is ensured since in both cases p's lock is held.
          */
         while (cur->retransid > -1 && ast_sched_del(sched, cur->retransid)) {
            sip_pvt_unlock(p);
            usleep(1);
            sip_pvt_lock(p);
         }
         UNLINK(cur, p->packets, prev);
         dialog_unref(cur->owner, "unref pkt cur->owner dialog from sip ack before freeing pkt");
         if (cur->data)
            ast_free(cur->data);
         ast_free(cur);
         break;
      }
   }
   ast_debug(1, "Stopping retransmission on '%s' of %s %d: Match %s\n",
      p->callid, resp ? "Response" : "Request", seqno, msg);
   return res;
}

static int __sip_autodestruct

(

const void *

data

)

[static]

Kill a SIP dialog (called only by the scheduler) The scheduler has a reference to this dialog when p->autokillid != -1, and we are called using that reference. So if the event is not rescheduled, we need to call dialog_unref().

Definition at line 4204 of file chan_sip.c.

References __sip_pretend_ack(), sip_pvt::alreadygone, append_history, AST_CAUSE_PROTOCOL_ERROR, ast_channel_trylock, ast_channel_unlock, ast_debug, AST_EXTENSION_DEACTIVATED, ast_log(), ast_queue_hangup_with_cause(), sip_pvt::autokillid, sip_pvt::callid, DEFAULT_TRANS_TIMEOUT, dialog_unlink_all(), dialog_unref(), sip_pvt::lastmsg, LOG_WARNING, sip_pvt::method, method_match(), sip_pvt::needdestroy, NONE, sip_pvt::owner, sip_pvt::packets, pvt_set_needdestroy(), sip_pvt::refer, SIP_BYE, SIP_CANCEL, sip_methods, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), sip_pvt::subscribed, cfsip_methods::text, transmit_request_with_auth(), transmit_state_notify(), TRUE, and XMIT_RELIABLE.

Referenced by sip_scheddestroy(), and sip_show_sched().

{
   struct sip_pvt *p = (struct sip_pvt *)data;

   /* If this is a subscription, tell the phone that we got a timeout */
   if (p->subscribed) {
      transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, TRUE);  /* Send last notification */
      p->subscribed = NONE;
      append_history(p, "Subscribestatus", "timeout");
      ast_debug(3, "Re-scheduled destruction of SIP subscription %s\n", p->callid ? p->callid : "<unknown>");
      return 10000;  /* Reschedule this destruction so that we know that it's gone */
   }

   /* If there are packets still waiting for delivery, delay the destruction */
   if (p->packets) {
      if (!p->needdestroy) {
         char method_str[31];
         ast_debug(3, "Re-scheduled destruction of SIP call %s\n", p->callid ? p->callid : "<unknown>");
         append_history(p, "ReliableXmit", "timeout");
         if (sscanf(p->lastmsg, "Tx: %30s", method_str) == 1 || sscanf(p->lastmsg, "Rx: %30s", method_str) == 1) {
            if (method_match(SIP_CANCEL, method_str) || method_match(SIP_BYE, method_str)) {
               pvt_set_needdestroy(p, "autodestruct");
            }
         }
         return 10000;
      } else {
         /* They've had their chance to respond. Time to bail */
         __sip_pretend_ack(p);
      }
   }

   /* Reset schedule ID */
   p->autokillid = -1;


   /*
    * Lock both the pvt and the channel safely so that we can queue up a frame.
    */
   sip_pvt_lock(p);
   while (p->owner && ast_channel_trylock(p->owner)) {
      sip_pvt_unlock(p);
      sched_yield();
      sip_pvt_lock(p);
   }

   if (p->owner) {
      ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
      ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
      ast_channel_unlock(p->owner);
   } else if (p->refer && !p->alreadygone) {
      ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
      transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
      append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   } else {
      append_history(p, "AutoDestroy", "%s", p->callid);
      ast_debug(3, "Auto destroying SIP dialog '%s'\n", p->callid);
      dialog_unlink_all(p, TRUE, TRUE); /* once it's unlinked and unrefd everywhere, it'll be freed automagically */
      /* dialog_unref(p, "unref dialog-- no other matching conditions"); -- unlink all now should finish off the dialog's references and free it. */
      /* sip_destroy(p); */      /* Go ahead and destroy dialog. All attempts to recover is done */
      /* sip_destroy also absorbs the reference */
   }

   sip_pvt_unlock(p);

   dialog_unref(p, "The ref to a dialog passed to this sched callback is going out of scope; unref it.");

   return 0;
}

static void __sip_destroy	(	struct sip_pvt *	p,
		int	lockowner,
		int	lockdialoglist
	)			[static]

Execute destruction of SIP dialog structure, release memory.

Definition at line 5791 of file chan_sip.c.

References ast_channel::_softhangup, ao2_ref, ast_channel_lock, ast_channel_unlock, ast_debug, ast_free, AST_LIST_REMOVE_HEAD, ast_log(), ast_rtp_destroy(), ast_rtp_get_bridged(), AST_SOFTHANGUP_DEV, ast_string_field_free_memory, ast_test_flag, ast_udptl_destroy(), ast_variables_destroy(), ast_verbose, sip_subscription_mwi::call, sip_registry::call, sip_peer::call, sip_pvt::callid, sip_pvt::chanvars, DEC_CALL_LIMIT, deinit_req(), dialog_unref(), sip_pvt::flags, free_old_route(), sip_pvt::history, sip_pvt::history_entries, sip_pvt::initreq, LOG_DEBUG, sip_pvt::method, sip_pvt::mwi, sip_peer::mwipvt, ast_channel::name, sip_pvt::notify_headers, option_debug, sip_pvt::options, sip_pvt::owner, sip_st_dlg::quit_flag, sip_pvt::refer, sip_pvt::registry, registry_unref(), sip_pvt::relatedpeer, sip_pvt::request_queue, sip_pvt::route, sip_pvt::rtp, sip_debug_test_pvt(), sip_dump_history(), SIP_INC_COUNT, sip_methods, SIP_PAGE2_CALL_ONHOLD, sip_pvt::socket, sip_pvt::stimer, stop_session_timer(), sip_socket::tcptls_session, ast_channel::tech_pvt, cfsip_methods::text, sip_pvt::trtp, sip_pvt::udptl, unref_peer(), update_call_counter(), and sip_pvt::vrtp.

Referenced by sip_destroy().

{
   struct sip_request *req;

   /* Destroy Session-Timers if allocated */
   if (p->stimer) {
      p->stimer->quit_flag = 1;
      stop_session_timer(p);
      ast_free(p->stimer);
      p->stimer = NULL;
   }

   if (sip_debug_test_pvt(p))
      ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);

   if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
      update_call_counter(p, DEC_CALL_LIMIT);
      ast_debug(2, "This call did not properly clean up call limits. Call ID %s\n", p->callid);
   }

   /* Unlink us from the owner if we have one */
   if (p->owner) {
      if (lockowner)
         ast_channel_lock(p->owner);
      if (option_debug)
         ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
      p->owner->tech_pvt = NULL;
      /* Make sure that the channel knows its backend is going away */
      p->owner->_softhangup |= AST_SOFTHANGUP_DEV;
      if (lockowner)
         ast_channel_unlock(p->owner);
      /* Give the channel a chance to react before deallocation */
      usleep(1);
   }

   /* Remove link from peer to subscription of MWI */
   if (p->relatedpeer && p->relatedpeer->mwipvt)
      p->relatedpeer->mwipvt = dialog_unref(p->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
   if (p->relatedpeer && p->relatedpeer->call == p)
      p->relatedpeer->call = dialog_unref(p->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
   
   if (p->relatedpeer)
      p->relatedpeer = unref_peer(p->relatedpeer,"unsetting a dialog relatedpeer field in sip_destroy");
   
   if (p->registry) {
      if (p->registry->call == p)
         p->registry->call = dialog_unref(p->registry->call, "nulling out the registry's call dialog field in unlink_all");
      p->registry = registry_unref(p->registry, "delete p->registry");
   }
   
   if (p->mwi) {
      p->mwi->call = NULL;
   }

   if (dumphistory)
      sip_dump_history(p);

   if (p->options)
      ast_free(p->options);

   if (p->notify_headers) {
      ast_variables_destroy(p->notify_headers);
      p->notify_headers = NULL;
   }
   if (p->rtp) {
      ast_rtp_destroy(p->rtp);
   }
   if (p->vrtp) {
      ast_rtp_destroy(p->vrtp);
   }
   if (p->trtp) {
      while (ast_rtp_get_bridged(p->trtp))
         usleep(1);
      ast_rtp_destroy(p->trtp);
   }
   if (p->udptl)
      ast_udptl_destroy(p->udptl);
   if (p->refer)
      ast_free(p->refer);
   if (p->route) {
      free_old_route(p->route);
      p->route = NULL;
   }
   deinit_req(&p->initreq);

   /* Clear history */
   if (p->history) {
      struct sip_history *hist;
      while ( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) ) {
         ast_free(hist);
         p->history_entries--;
      }
      ast_free(p->history);
      p->history = NULL;
   }

   while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) {
      ast_free(req);
   }

   if (p->chanvars) {
      ast_variables_destroy(p->chanvars);
      p->chanvars = NULL;
   }

   ast_string_field_free_memory(p);

   if (p->socket.tcptls_session) {
      ao2_ref(p->socket.tcptls_session, -1);
      p->socket.tcptls_session = NULL;
   }
}

static int __sip_do_register

(

struct sip_registry *

r

)

[static]

Register with SIP proxy.

Definition at line 12073 of file chan_sip.c.

References SIP_REGISTER, and transmit_register().

Referenced by sip_reregister().

{
   int res;

   res = transmit_register(r, SIP_REGISTER, NULL, NULL);
   return res;
}

static void __sip_pretend_ack

(

struct sip_pvt *

p

)

[static]

Pretend to ack all packets called with p locked.

Definition at line 4377 of file chan_sip.c.

References __sip_ack(), ast_log(), sip_pkt::data, find_sip_method(), sip_pkt::is_resp, LOG_WARNING, sip_pkt::method, sip_request::method, sip_pvt::packets, sip_pkt::seqno, sip_methods, and cfsip_methods::text.

Referenced by __sip_autodestruct(), handle_request_bye(), handle_request_cancel(), and sip_reg_timeout().

{
   struct sip_pkt *cur = NULL;

   while (p->packets) {
      int method;
      if (cur == p->packets) {
         ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
         return;
      }
      cur = p->packets;
      method = (cur->method) ? cur->method : find_sip_method(cur->data->str);
      __sip_ack(p, cur->seqno, cur->is_resp, method);
   }
}

static enum sip_result __sip_reliable_xmit	(	struct sip_pvt *	p,
		int	seqno,
		int	resp,
		struct ast_str *	data,
		int	len,
		int	fatal,
		int	sipmethod
	)			[static]

Transmit packet with retransmits.

Returns:

0 on success, -1 on failure to allocate packet

Todo:

According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited

Definition at line 4125 of file chan_sip.c.

References __sip_xmit(), append_history, ast_calloc, ast_debug, AST_FAILURE, ast_free, ast_log(), AST_SCHED_DEL, AST_SCHED_REPLACE_VARIABLE, ast_str_buffer(), ast_str_create(), ast_str_set(), AST_SUCCESS, sip_pkt::data, DEFAULT_RETRANS, dialog_ref(), dialog_unref(), sip_pkt::is_fatal, sip_pkt::is_resp, LOG_ERROR, sip_pkt::method, sip_pkt::next, sip_pkt::owner, sip_pkt::packetlen, sip_pvt::packets, sip_pvt::pendinginvite, sip_pkt::response_code, retrans_pkt(), sip_pkt::retransid, sip_pkt::seqno, SIP_INVITE, SIP_TRANSPORT_UDP, sip_pvt::socket, sip_pvt::timer_t1, sip_pkt::timer_t1, sip_socket::type, and XMIT_ERROR.

Referenced by send_request(), and send_response().

{
   struct sip_pkt *pkt = NULL;
   int siptimer_a = DEFAULT_RETRANS;
   int xmitres = 0;
   int respid;

   if (sipmethod == SIP_INVITE) {
      /* Note this is a pending invite */
      p->pendinginvite = seqno;
   }

   /* If the transport is something reliable (TCP or TLS) then don't really send this reliably */
   /* I removed the code from retrans_pkt that does the same thing so it doesn't get loaded into the scheduler */
   /*! \todo According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited */
   if (!(p->socket.type & SIP_TRANSPORT_UDP)) {
      xmitres = __sip_xmit(p, data, len); /* Send packet */
      if (xmitres == XMIT_ERROR) {  /* Serious network trouble, no need to try again */
         append_history(p, "XmitErr", "%s", fatal ? "(Critical)" : "(Non-critical)");
         return AST_FAILURE;
      } else {
         return AST_SUCCESS;
      }
   }

   if (!(pkt = ast_calloc(1, sizeof(*pkt))))
      return AST_FAILURE;
   /* copy data, add a terminator and save length */
   if (!(pkt->data = ast_str_create(len))) {
      ast_free(pkt);
      return AST_FAILURE;
   }
   ast_str_set(&pkt->data, 0, "%s%s", data->str, "\0");
   pkt->packetlen = len;
   /* copy other parameters from the caller */
   pkt->method = sipmethod;
   pkt->seqno = seqno;
   pkt->is_resp = resp;
   pkt->is_fatal = fatal;
   pkt->owner = dialog_ref(p, "__sip_reliable_xmit: setting pkt->owner");
   pkt->next = p->packets;
   p->packets = pkt; /* Add it to the queue */
   if (resp) {
      /* Parse out the response code */
      if (sscanf(ast_str_buffer(pkt->data), "SIP/2.0 %30u", &respid) == 1) {
         pkt->response_code = respid;
      }
   }
   pkt->timer_t1 = p->timer_t1;  /* Set SIP timer T1 */
   pkt->retransid = -1;
   if (pkt->timer_t1)
      siptimer_a = pkt->timer_t1 * 2;

   /* Schedule retransmission */
   AST_SCHED_REPLACE_VARIABLE(pkt->retransid, sched, siptimer_a, retrans_pkt, pkt, 1);
   if (sipdebug)
      ast_debug(4, "*** SIP TIMER: Initializing retransmit timer on packet: Id  #%d\n", pkt->retransid);

   xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);   /* Send packet */

   if (xmitres == XMIT_ERROR) {  /* Serious network trouble, no need to try again */
      append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
      ast_log(LOG_ERROR, "Serious Network Trouble; __sip_xmit returns error for pkt data\n");
      AST_SCHED_DEL(sched, pkt->retransid);
      p->packets = pkt->next;
      pkt->owner = dialog_unref(pkt->owner,"pkt is being freed, its dialog ref is dead now");
      ast_free(pkt->data);
      ast_free(pkt);
      return AST_FAILURE;
   } else {
      return AST_SUCCESS;
   }
}

static int __sip_semi_ack	(	struct sip_pvt *	p,
		int	seqno,
		int	resp,
		int	sipmethod
	)			[static]

Acks receipt of packet, keep it around (used for provisional responses).

Definition at line 4394 of file chan_sip.c.

References ast_debug, AST_SCHED_DEL, sip_pvt::callid, sip_pkt::data, FALSE, sip_pkt::is_resp, method_match(), sip_pkt::next, sip_pvt::packets, sip_pkt::retransid, sip_pkt::seqno, sip_methods, cfsip_methods::text, and TRUE.

Referenced by handle_response(), and sip_hangup().

{
   struct sip_pkt *cur;
   int res = FALSE;

   for (cur = p->packets; cur; cur = cur->next) {
      if (cur->seqno == seqno && cur->is_resp == resp &&
         (cur->is_resp || method_match(sipmethod, cur->data->str))) {
         /* this is our baby */
         if (cur->retransid > -1) {
            if (sipdebug)
               ast_debug(4, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
         }
         AST_SCHED_DEL(sched, cur->retransid);
         res = TRUE;
         break;
      }
   }
   ast_debug(1, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res == -1 ? "Not Found" : "Found");
   return res;
}

static int __sip_subscribe_mwi_do

(

struct sip_subscription_mwi *

mwi

)

[static]

Actually setup an MWI subscription or resubscribe.

Definition at line 11554 of file chan_sip.c.

References ast_dnsmgr_lookup(), ast_set_flag, ast_sip_ouraddrfor(), ast_string_field_set, ast_strlen_zero(), ASTOBJ_REF, sip_subscription_mwi::authuser, build_callid_pvt(), build_contact(), build_via(), sip_subscription_mwi::call, create_addr(), dialog_unlink_all(), dialog_unref(), sip_subscription_mwi::dnsmgr, sip_pvt::flags, get_srv_protocol(), get_srv_service(), sip_subscription_mwi::hostname, MAXHOSTNAMELEN, sip_pvt::mwi, obproxy_get(), sip_pvt::ourip, sip_socket::port, sip_subscription_mwi::portno, sip_pvt::recv, ref_proxy(), sip_pvt::sa, sip_subscription_mwi::secret, set_socket_transport(), sip_alloc(), sip_cfg, SIP_OUTGOING, SIP_SUBSCRIBE, sip_pvt::socket, sip_settings::srvlookup, transmit_invite(), sip_subscription_mwi::transport, TRUE, sip_subscription_mwi::us, and sip_subscription_mwi::username.

Referenced by sip_subscribe_mwi_do().

{
   /* If we have no DNS manager let's do a lookup */
   if (!mwi->dnsmgr) {
      char transport[MAXHOSTNAMELEN];
      snprintf(transport, sizeof(transport), "_%s._%s", get_srv_service(mwi->transport), get_srv_protocol(mwi->transport));
      ast_dnsmgr_lookup(mwi->hostname, &mwi->us, &mwi->dnsmgr, sip_cfg.srvlookup ? transport : NULL);
   }

   /* If we already have a subscription up simply send a resubscription */
   if (mwi->call) {
      transmit_invite(mwi->call, SIP_SUBSCRIBE, 0, 0);
      return 0;
   }
   
   /* Create a dialog that we will use for the subscription */
   if (!(mwi->call = sip_alloc(NULL, NULL, 0, SIP_SUBSCRIBE, NULL))) {
      return -1;
   }

   ref_proxy(mwi->call, obproxy_get(mwi->call, NULL));

   if (!mwi->us.sin_port && mwi->portno) {
      mwi->us.sin_port = htons(mwi->portno);
   }
   
   /* Setup the destination of our subscription */
   if (create_addr(mwi->call, mwi->hostname, &mwi->us, 0)) {
      dialog_unlink_all(mwi->call, TRUE, TRUE);
      mwi->call = dialog_unref(mwi->call, "unref dialog after unlink_all");
      return 0;
   }
   
   if (!mwi->dnsmgr && mwi->portno) {
      mwi->call->sa.sin_port = htons(mwi->portno);
      mwi->call->recv.sin_port = htons(mwi->portno);
   } else {
      mwi->portno = ntohs(mwi->call->sa.sin_port);
   }
   
   /* Set various other information */
   if (!ast_strlen_zero(mwi->authuser)) {
      ast_string_field_set(mwi->call, peername, mwi->authuser);
      ast_string_field_set(mwi->call, authname, mwi->authuser);
      ast_string_field_set(mwi->call, fromuser, mwi->authuser);
   } else {
      ast_string_field_set(mwi->call, peername, mwi->username);
      ast_string_field_set(mwi->call, authname, mwi->username);
      ast_string_field_set(mwi->call, fromuser, mwi->username);
   }
   ast_string_field_set(mwi->call, username, mwi->username);
   if (!ast_strlen_zero(mwi->secret)) {
      ast_string_field_set(mwi->call, peersecret, mwi->secret);
   }
   set_socket_transport(&mwi->call->socket, mwi->transport);
   mwi->call->socket.port = htons(mwi->portno);
   ast_sip_ouraddrfor(&mwi->call->sa.sin_addr, &mwi->call->ourip, mwi->call);
   build_contact(mwi->call);
   build_via(mwi->call);
   build_callid_pvt(mwi->call);
   ast_set_flag(&mwi->call->flags[0], SIP_OUTGOING);
   
   /* Associate the call with us */
   mwi->call->mwi = ASTOBJ_REF(mwi);
   
   /* Actually send the packet */
   transmit_invite(mwi->call, SIP_SUBSCRIBE, 0, 2);

   return 0;
}

static int __sip_xmit	(	struct sip_pvt *	p,
		struct ast_str *	data,
		int	len
	)			[static]

Transmit SIP message Sends a SIP request or response on a given socket (in the pvt) Called by retrans_pkt, send_request, send_response and __sip_reliable_xmit.

Definition at line 3822 of file chan_sip.c.

References ast_debug, ast_inet_ntoa(), ast_log(), errno, sip_socket::fd, get_transport_pvt(), LOG_WARNING, sip_prepare_socket(), sip_real_dst(), sip_tcptls_write(), SIP_TRANSPORT_UDP, sip_pvt::socket, sip_socket::tcptls_session, sip_socket::type, and XMIT_ERROR.

Referenced by __sip_reliable_xmit(), retrans_pkt(), send_request(), and send_response().

{
   int res = 0;
   const struct sockaddr_in *dst = sip_real_dst(p);

   ast_debug(2, "Trying to put '%.11s' onto %s socket destined for %s:%d\n", data->str, get_transport_pvt(p), ast_inet_ntoa(dst->sin_addr), htons(dst->sin_port));

   if (sip_prepare_socket(p) < 0)
      return XMIT_ERROR;

   if (p->socket.type == SIP_TRANSPORT_UDP) {
      res = sendto(p->socket.fd, data->str, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
   } else if (p->socket.tcptls_session) {
      res = sip_tcptls_write(p->socket.tcptls_session, data->str, len);
   } else {
      ast_debug(2, "Socket type is TCP but no tcptls_session is present to write to\n");
      return XMIT_ERROR;
   }

   if (res == -1) {
      switch (errno) {
      case EBADF:       /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
      case EHOSTUNREACH:   /* Host can't be reached */
      case ENETDOWN:       /* Inteface down */
      case ENETUNREACH: /* Network failure */
      case ECONNREFUSED:      /* ICMP port unreachable */ 
         res = XMIT_ERROR; /* Don't bother with trying to transmit again */
      }
   }
   if (res != len)
      ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));

   return res;
}

static int __transmit_response	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		enum xmittype	reliable
	)			[static]

Base transmit response function.

Definition at line 10156 of file chan_sip.c.

References add_header(), ast_cause2str(), ast_log(), buf, get_header(), ast_channel::hangupcause, LOG_WARNING, sip_pvt::method, sip_pvt::owner, respprep(), send_response(), and SIP_INVITE.

Referenced by transmit_response(), transmit_response_reliable(), and transmit_response_using_temp().

{
   struct sip_request resp;
   int seqno = 0;

   if (reliable && (sscanf(get_header(req, "CSeq"), "%30d ", &seqno) != 1)) {
      ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq"));
      return -1;
   }
   respprep(&resp, p, msg, req);
   /* If we are cancelling an incoming invite for some reason, add information
      about the reason why we are doing this in clear text */
   if (p->method == SIP_INVITE && msg[0] != '1' && p->owner && p->owner->hangupcause) {
      char buf[10];

      add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause));
      snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause);
      add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
   }
   return send_response(p, &resp, reliable, seqno);
}

static void __unreg_module

(

void

)

[static]

Definition at line 26863 of file chan_sip.c.

static char * _sip_qualify_peer	(	int	type,
		int	fd,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)			[static]

Send qualify message to peer from cli or manager. Mostly for debugging.

Definition at line 15925 of file chan_sip.c.

References ast_cli(), astman_send_error(), CLI_SHOWUSAGE, CLI_SUCCESS, FALSE, find_peer(), FINDPEERS, sip_poke_peer(), TRUE, and unref_peer().

Referenced by manager_sip_qualify_peer(), and sip_qualify_peer().

{
   struct sip_peer *peer;
   int load_realtime;

   if (argc < 4)
      return CLI_SHOWUSAGE;

   load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
   if ((peer = find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE, 0))) {
      sip_poke_peer(peer, 1);
      unref_peer(peer, "qualify: done with peer");
   } else if (type == 0) {
      ast_cli(fd, "Peer '%s' not found\n", argv[3]);
   } else {
      astman_send_error(s, m, "Peer not found");
   }
   return CLI_SUCCESS;
}

static char * _sip_show_peer	(	int	type,
		int	fd,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)			[static]

Show one peer in detail (main function).

Definition at line 16010 of file chan_sip.c.

References sip_peer::accountcode, sip_peer::addr, sip_peer::allowtransfer, sip_peer::amaflags, ARRAY_LEN, ast_callerid_merge(), ast_cdr_flags2str(), ast_check_realtime(), ast_cli(), ast_codec_pref_index(), ast_describe_caller_presentation(), ast_getformatname(), ast_getformatname_multiple(), ast_inet_ntoa(), ast_print_group(), ast_sched_when(), ast_str_alloca, ast_strlen_zero(), ast_test_flag, astman_append(), astman_get_header(), astman_send_error(), sip_peer::auth, sip_peer::autoframing, sip_peer::busy_level, sip_peer::call_limit, sip_peer::callgroup, sip_peer::callingpres, sip_peer::capability, sip_peer::chanvars, sip_peer::cid_name, sip_peer::cid_num, CLI_SHOWUSAGE, CLI_SUCCESS, cli_yesno(), sip_peer::context, sip_peer::defaddr, dtmfmode2str(), sip_peer::expire, FALSE, faxec2str(), find_peer(), FINDPEERS, sip_peer::flags, sip_proxy::force, sip_peer::fromdomain, sip_peer::fromuser, sip_peer::fullcontact, get_transport(), get_transport_list(), sip_peer::ha, sip_peer::host_dynamic, cfsip_options::id, insecure2str(), sip_peer::is_realtime, sip_peer::language, sip_peer::lastmsgssent, sip_peer::maxcallbitrate, sip_auth::md5secret, sip_peer::md5secret, ast_variable::name, sip_proxy::name, sip_peer::name, nat2str(), ast_variable::next, sip_auth::next, sip_peer::outboundproxy, sip_peer::parkinglot, peer_mailboxes_to_str(), peer_status(), sip_peer::pickupgroup, sip_peer::prefs, print_codec_to_cli(), print_group(), sip_peer::qualifyfreq, sip_auth::realm, sip_peer::regexten, sip_peer::remotesecret, S_OR, sip_auth::secret, sip_peer::secret, SIP_DIRECT_MEDIA, SIP_DTMF, SIP_INSECURE, SIP_NAT, sip_options, SIP_PAGE2_ALLOWOVERLAP, SIP_PAGE2_ALLOWSUBSCRIBE, SIP_PAGE2_FORWARD_LOOP_DETECTED, SIP_PAGE2_IGNORESDPVERSION, SIP_PAGE2_REGISTERTRYING, SIP_PAGE2_T38SUPPORT, SIP_PAGE2_TEXTSUPPORT, SIP_PAGE2_VIDEOSUPPORT, SIP_PAGE2_VIDEOSUPPORT_ALWAYS, SIP_PROMISCREDIR, SIP_SENDRPID, SIP_TRUSTRPID, SIP_USEREQPHONE, sip_peer::sipoptions, sip_peer::socket, sip_st_cfg::st_max_se, sip_st_cfg::st_min_se, sip_st_cfg::st_mode_oper, sip_st_cfg::st_ref, status, sip_peer::stimer, stmode2str(), strefresher2str(), sip_peer::subscribecontext, sip_peer::t38_maxdatagram, text, sip_peer::timer_b, sip_peer::timer_t1, sip_peer::tohost, transfermode2str(), sip_peer::transports, TRUE, sip_socket::type, unref_peer(), sip_peer::useragent, sip_peer::username, sip_auth::username, ast_variable::value, and sip_peer::vmexten.

Referenced by manager_sip_show_peer(), and sip_show_peer().

{
   char status[30] = "";
   char cbuf[256];
   struct sip_peer *peer;
   char codec_buf[512];
   struct ast_codec_pref *pref;
   struct ast_variable *v;
   struct sip_auth *auth;
   int x = 0, codec = 0, load_realtime;
   int realtimepeers;

   realtimepeers = ast_check_realtime("sippeers");

   if (argc < 4)
      return CLI_SHOWUSAGE;

   load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
   peer = find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE, 0);

   if (s) {    /* Manager */
      if (peer) {
         const char *id = astman_get_header(m, "ActionID");

         astman_append(s, "Response: Success\r\n");
         if (!ast_strlen_zero(id))
            astman_append(s, "ActionID: %s\r\n", id);
      } else {
         snprintf (cbuf, sizeof(cbuf), "Peer %s not found.", argv[3]);
         astman_send_error(s, m, cbuf);
         return CLI_SUCCESS;
      }
   }
   if (peer && type==0 ) { /* Normal listing */
      struct ast_str *mailbox_str = ast_str_alloca(512);
      ast_cli(fd, "\n\n");
      ast_cli(fd, "  * Name       : %s\n", peer->name);
      if (realtimepeers) { /* Realtime is enabled */
         ast_cli(fd, "  Realtime peer: %s\n", peer->is_realtime ? "Yes, cached" : "No");
      }
      ast_cli(fd, "  Secret       : %s\n", ast_strlen_zero(peer->secret)?"<Not set>":"<Set>");
      ast_cli(fd, "  MD5Secret    : %s\n", ast_strlen_zero(peer->md5secret)?"<Not set>":"<Set>");
      ast_cli(fd, "  Remote Secret: %s\n", ast_strlen_zero(peer->remotesecret)?"<Not set>":"<Set>");
      for (auth = peer->auth; auth; auth = auth->next) {
         ast_cli(fd, "  Realm-auth   : Realm %-15.15s User %-10.20s ", auth->realm, auth->username);
         ast_cli(fd, "%s\n", !ast_strlen_zero(auth->secret)?"<Secret set>":(!ast_strlen_zero(auth->md5secret)?"<MD5secret set>" : "<Not set>"));
      }
      ast_cli(fd, "  Context      : %s\n", peer->context);
      ast_cli(fd, "  Subscr.Cont. : %s\n", S_OR(peer->subscribecontext, "<Not set>") );
      ast_cli(fd, "  Language     : %s\n", peer->language);
      if (!ast_strlen_zero(peer->accountcode))
         ast_cli(fd, "  Accountcode  : %s\n", peer->accountcode);
      ast_cli(fd, "  AMA flags    : %s\n", ast_cdr_flags2str(peer->amaflags));
      ast_cli(fd, "  Transfer mode: %s\n", transfermode2str(peer->allowtransfer));
      ast_cli(fd, "  CallingPres  : %s\n", ast_describe_caller_presentation(peer->callingpres));
      if (!ast_strlen_zero(peer->fromuser))
         ast_cli(fd, "  FromUser     : %s\n", peer->fromuser);
      if (!ast_strlen_zero(peer->fromdomain))
         ast_cli(fd, "  FromDomain   : %s\n", peer->fromdomain);
      ast_cli(fd, "  Callgroup    : ");
      print_group(fd, peer->callgroup, 0);
      ast_cli(fd, "  Pickupgroup  : ");
      print_group(fd, peer->pickupgroup, 0);
      peer_mailboxes_to_str(&mailbox_str, peer);
      ast_cli(fd, "  Mailbox      : %s\n", mailbox_str->str);
      ast_cli(fd, "  VM Extension : %s\n", peer->vmexten);
      ast_cli(fd, "  LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff);
      ast_cli(fd, "  Call limit   : %d\n", peer->call_limit);
      if (peer->busy_level)
         ast_cli(fd, "  Busy level   : %d\n", peer->busy_level);
      ast_cli(fd, "  Dynamic      : %s\n", cli_yesno(peer->host_dynamic));
      ast_cli(fd, "  Callerid     : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>"));
      ast_cli(fd, "  MaxCallBR    : %d kbps\n", peer->maxcallbitrate);
      ast_cli(fd, "  Expire       : %ld\n", ast_sched_when(sched, peer->expire));
      ast_cli(fd, "  Insecure     : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
      ast_cli(fd, "  Nat          : %s\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
      ast_cli(fd, "  ACL          : %s\n", cli_yesno(peer->ha != NULL));
      ast_cli(fd, "  T.38 support : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
      ast_cli(fd, "  T.38 EC mode : %s\n", faxec2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
      ast_cli(fd, "  T.38 MaxDtgrm: %d\n", peer->t38_maxdatagram);
      ast_cli(fd, "  DirectMedia  : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)));
      ast_cli(fd, "  PromiscRedir : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)));
      ast_cli(fd, "  User=Phone   : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
      ast_cli(fd, "  Video Support: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) || ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS)));
      ast_cli(fd, "  Text Support : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)));
      ast_cli(fd, "  Ign SDP ver  : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_IGNORESDPVERSION)));
      ast_cli(fd, "  Trust RPID   : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_TRUSTRPID)));
      ast_cli(fd, "  Send RPID    : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_SENDRPID)));
      ast_cli(fd, "  Subscriptions: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
      ast_cli(fd, "  Overlap dial : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
      ast_cli(fd, "  Forward Loop : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED)));
      if (peer->outboundproxy)
         ast_cli(fd, "  Outb. proxy  : %s %s\n", ast_strlen_zero(peer->outboundproxy->name) ? "<not set>" : peer->outboundproxy->name,
                     peer->outboundproxy->force ? "(forced)" : "");

      /* - is enumerated */
      ast_cli(fd, "  DTMFmode     : %s\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
      ast_cli(fd, "  Timer T1     : %d\n", peer->timer_t1);
      ast_cli(fd, "  Timer B      : %d\n", peer->timer_b);
      ast_cli(fd, "  ToHost       : %s\n", peer->tohost);
      ast_cli(fd, "  Addr->IP     : %s Port %d\n",  peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port));
      ast_cli(fd, "  Defaddr->IP  : %s Port %d\n", ast_inet_ntoa(peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
      ast_cli(fd, "  Prim.Transp. : %s\n", get_transport(peer->socket.type));
      ast_cli(fd, "  Allowed.Trsp : %s\n", get_transport_list(peer->transports)); 
      if (!ast_strlen_zero(global_regcontext))
         ast_cli(fd, "  Reg. exten   : %s\n", peer->regexten);
      ast_cli(fd, "  Def. Username: %s\n", peer->username);
      ast_cli(fd, "  SIP Options  : ");
      if (peer->sipoptions) {
         int lastoption = -1;
         for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
            if (sip_options[x].id != lastoption) {
               if (peer->sipoptions & sip_options[x].id)
                  ast_cli(fd, "%s ", sip_options[x].text);
               lastoption = x;
            }
         }
      } else
         ast_cli(fd, "(none)");

      ast_cli(fd, "\n");
      ast_cli(fd, "  Codecs       : ");
      ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
      ast_cli(fd, "%s\n", codec_buf);
      ast_cli(fd, "  Codec Order  : (");
      print_codec_to_cli(fd, &peer->prefs);
      ast_cli(fd, ")\n");

      ast_cli(fd, "  Auto-Framing :  %s \n", cli_yesno(peer->autoframing));
      ast_cli(fd, "  100 on REG   : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_REGISTERTRYING) ? "Yes" : "No");
      ast_cli(fd, "  Status       : ");
      peer_status(peer, status, sizeof(status));
      ast_cli(fd, "%s\n", status);
      ast_cli(fd, "  Useragent    : %s\n", peer->useragent);
      ast_cli(fd, "  Reg. Contact : %s\n", peer->fullcontact);
      ast_cli(fd, "  Qualify Freq : %d ms\n", peer->qualifyfreq);
      if (peer->chanvars) {
         ast_cli(fd, "  Variables    :\n");
         for (v = peer->chanvars ; v ; v = v->next)
            ast_cli(fd, "                 %s = %s\n", v->name, v->value);
      }

      ast_cli(fd, "  Sess-Timers  : %s\n", stmode2str(peer->stimer.st_mode_oper));
      ast_cli(fd, "  Sess-Refresh : %s\n", strefresher2str(peer->stimer.st_ref));
      ast_cli(fd, "  Sess-Expires : %d secs\n", peer->stimer.st_max_se);
      ast_cli(fd, "  Min-Sess     : %d secs\n", peer->stimer.st_min_se);
      ast_cli(fd, "  Parkinglot   : %s\n", peer->parkinglot);
      ast_cli(fd, "\n");
      peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer ptr");
   } else  if (peer && type == 1) { /* manager listing */
      char buffer[256];
      struct ast_str *mailbox_str = ast_str_alloca(512);
      astman_append(s, "Channeltype: SIP\r\n");
      astman_append(s, "ObjectName: %s\r\n", peer->name);
      astman_append(s, "ChanObjectType: peer\r\n");
      astman_append(s, "SecretExist: %s\r\n", ast_strlen_zero(peer->secret)?"N":"Y");
      astman_append(s, "RemoteSecretExist: %s\r\n", ast_strlen_zero(peer->remotesecret)?"N":"Y");
      astman_append(s, "MD5SecretExist: %s\r\n", ast_strlen_zero(peer->md5secret)?"N":"Y");
      astman_append(s, "Context: %s\r\n", peer->context);
      astman_append(s, "Language: %s\r\n", peer->language);
      if (!ast_strlen_zero(peer->accountcode))
         astman_append(s, "Accountcode: %s\r\n", peer->accountcode);
      astman_append(s, "AMAflags: %s\r\n", ast_cdr_flags2str(peer->amaflags));
      astman_append(s, "CID-CallingPres: %s\r\n", ast_describe_caller_presentation(peer->callingpres));
      if (!ast_strlen_zero(peer->fromuser))
         astman_append(s, "SIP-FromUser: %s\r\n", peer->fromuser);
      if (!ast_strlen_zero(peer->fromdomain))
         astman_append(s, "SIP-FromDomain: %s\r\n", peer->fromdomain);
      astman_append(s, "Callgroup: ");
      astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->callgroup));
      astman_append(s, "Pickupgroup: ");
      astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->pickupgroup));
      peer_mailboxes_to_str(&mailbox_str, peer);
      astman_append(s, "VoiceMailbox: %s\r\n", mailbox_str->str);
      astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer));
      astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
      astman_append(s, "Call-limit: %d\r\n", peer->call_limit);
      astman_append(s, "Busy-level: %d\r\n", peer->busy_level);
      astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate);
      astman_append(s, "Dynamic: %s\r\n", peer->host_dynamic?"Y":"N");
      astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
      astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched, peer->expire));
      astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
      astman_append(s, "SIP-NatSupport: %s\r\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
      astman_append(s, "ACL: %s\r\n", (peer->ha?"Y":"N"));
      astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
      astman_append(s, "SIP-DirectMedia: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
      astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N"));
      astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N"));
      astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N"));
      astman_append(s, "SIP-TextSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Y":"N"));
      astman_append(s, "SIP-T.38Support: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)?"Y":"N"));
      astman_append(s, "SIP-T.38EC: %s\r\n", faxec2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
      astman_append(s, "SIP-T.38MaxDtgrm: %d\r\n", peer->t38_maxdatagram);
      astman_append(s, "SIP-Sess-Timers: %s\r\n", stmode2str(peer->stimer.st_mode_oper));
      astman_append(s, "SIP-Sess-Refresh: %s\r\n", strefresher2str(peer->stimer.st_ref));
      astman_append(s, "SIP-Sess-Expires: %d\r\n", peer->stimer.st_max_se);
      astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);

      /* - is enumerated */
      astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
      astman_append(s, "ToHost: %s\r\n", peer->tohost);
      astman_append(s, "Address-IP: %s\r\nAddress-Port: %d\r\n",  peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "", ntohs(peer->addr.sin_port));
      astman_append(s, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_inet_ntoa(peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
      astman_append(s, "Default-Username: %s\r\n", peer->username);
      if (!ast_strlen_zero(global_regcontext))
         astman_append(s, "RegExtension: %s\r\n", peer->regexten);
      astman_append(s, "Codecs: ");
      ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
      astman_append(s, "%s\r\n", codec_buf);
      astman_append(s, "CodecOrder: ");
      pref = &peer->prefs;
      for(x = 0; x < 32 ; x++) {
         codec = ast_codec_pref_index(pref, x);
         if (!codec)
            break;
         astman_append(s, "%s", ast_getformatname(codec));
         if (x < 31 && ast_codec_pref_index(pref, x+1))
            astman_append(s, ",");
      }

      astman_append(s, "\r\n");
      astman_append(s, "Status: ");
      peer_status(peer, status, sizeof(status));
      astman_append(s, "%s\r\n", status);
      astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent);
      astman_append(s, "Reg-Contact: %s\r\n", peer->fullcontact);
      astman_append(s, "QualifyFreq: %d ms\r\n", peer->qualifyfreq);
      astman_append(s, "Parkinglot: %s\r\n", peer->parkinglot);
      if (peer->chanvars) {
         for (v = peer->chanvars ; v ; v = v->next) {
            astman_append(s, "ChanVariable: %s=%s\r\n", v->name, v->value);
         }
      }

      peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer");

   } else {
      ast_cli(fd, "Peer %s not found.\n", argv[3]);
      ast_cli(fd, "\n");
   }

   return CLI_SUCCESS;
}

static char * _sip_show_peers	(	int	fd,
		int *	total,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)			[static]

Execute sip show peers command.

Definition at line 15327 of file chan_sip.c.

References sip_peer::addr, ao2_container_count(), ao2_iterator_destroy(), ao2_iterator_init(), ao2_lock(), ao2_t_iterator_next, ao2_unlock(), ast_calloc, ast_check_realtime(), ast_cli(), ast_copy_string(), ast_free, ast_inet_ntoa(), ast_strlen_zero(), ast_test_flag, astman_append(), astman_get_header(), CLI_SHOWUSAGE, CLI_SUCCESS, FALSE, sip_peer::flags, FORMAT, FORMAT2, sip_peer::ha, sip_peer::host_dynamic, id, sip_peer::is_realtime, sip_peer::name, name, peer_status(), peercomparefunc(), SIP_NAT_ROUTE, SIP_PAGE2_TEXTSUPPORT, SIP_PAGE2_VIDEOSUPPORT, SIP_TYPE_PEER, status, TRUE, sip_peer::type, unref_peer(), and sip_peer::username.

Referenced by manager_sip_show_peers(), and sip_show_peers().

{
   regex_t regexbuf;
   int havepattern = FALSE;
   struct sip_peer *peer;
   struct ao2_iterator i;
   
/* the last argument is left-aligned, so we don't need a size anyways */
#define FORMAT2 "%-25.25s  %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %s\n"
#define FORMAT  "%-25.25s  %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %s\n"

   char name[256];
   int total_peers = 0;
   int peers_mon_online = 0;
   int peers_mon_offline = 0;
   int peers_unmon_offline = 0;
   int peers_unmon_online = 0;
   const char *id;
   char idtext[256] = "";
   int realtimepeers;
   int objcount = ao2_container_count(peers);
   struct sip_peer **peerarray;
   int k;
   
   
   realtimepeers = ast_check_realtime("sippeers");
   peerarray = ast_calloc(sizeof(struct sip_peer *), objcount);

   if (s) { /* Manager - get ActionID */
      id = astman_get_header(m, "ActionID");
      if (!ast_strlen_zero(id))
         snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);
   }

   switch (argc) {
   case 5:
      if (!strcasecmp(argv[3], "like")) {
         if (regcomp(&regexbuf, argv[4], REG_EXTENDED | REG_NOSUB))
            return CLI_SHOWUSAGE;
         havepattern = TRUE;
      } else
         return CLI_SHOWUSAGE;
   case 3:
      break;
   default:
      return CLI_SHOWUSAGE;
   }

   if (!s) /* Normal list */
      ast_cli(fd, FORMAT2, "Name/username", "Host", "Dyn", "Nat", "ACL", "Port", "Status", (realtimepeers ? "Realtime" : ""));
   

   i = ao2_iterator_init(peers, 0);
   while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {  
      ao2_lock(peer);

      if (!(peer->type & SIP_TYPE_PEER)) {
         ao2_unlock(peer);
         unref_peer(peer, "unref peer because it's actually a user");
         continue;
      }

      if (havepattern && regexec(&regexbuf, peer->name, 0, NULL, 0)) {
         objcount--;
         ao2_unlock(peer);
         unref_peer(peer, "toss iterator peer ptr before continue");
         continue;
      }

      peerarray[total_peers++] = peer;
      ao2_unlock(peer);
   }
   ao2_iterator_destroy(&i);
   
   qsort(peerarray, total_peers, sizeof(struct sip_peer *), peercomparefunc);

   for(k=0; k < total_peers; k++) {
      char status[20] = "";
      char srch[2000];
      char pstatus;
      peer = peerarray[k];
      
      ao2_lock(peer);
      if (havepattern && regexec(&regexbuf, peer->name, 0, NULL, 0)) {
         ao2_unlock(peer);
         peer = peerarray[k] = unref_peer(peer, "toss iterator peer ptr before continue");
         continue;
      }

      if (!ast_strlen_zero(peer->username) && !s)
         snprintf(name, sizeof(name), "%s/%s", peer->name, peer->username);
      else
         ast_copy_string(name, peer->name, sizeof(name));
      
      pstatus = peer_status(peer, status, sizeof(status));
      if (pstatus == 1)
         peers_mon_online++;
      else if (pstatus == 0)
         peers_mon_offline++;
      else {
         if (peer->addr.sin_port == 0)
            peers_unmon_offline++;
         else
            peers_unmon_online++;
      }

      snprintf(srch, sizeof(srch), FORMAT, name,
         peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)",
         peer->host_dynamic ? " D " : "   ",    /* Dynamic or not? */
         ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE) ? " N " : "   ", /* NAT=yes? */
         peer->ha ? " A " : "   ",  /* permit/deny */
         ntohs(peer->addr.sin_port), status,
         realtimepeers ? (peer->is_realtime ? "Cached RT":"") : "");

      if (!s)  {/* Normal CLI list */
         ast_cli(fd, FORMAT, name, 
         peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)",
         peer->host_dynamic ? " D " : "   ",    /* Dynamic or not? */
         ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE) ? " N " : "   ", /* NAT=yes? */
         peer->ha ? " A " : "   ",       /* permit/deny */
         
         ntohs(peer->addr.sin_port), status,
         realtimepeers ? (peer->is_realtime ? "Cached RT":"") : "");
      } else { /* Manager format */
         /* The names here need to be the same as other channels */
         astman_append(s, 
         "Event: PeerEntry\r\n%s"
         "Channeltype: SIP\r\n"
         "ObjectName: %s\r\n"
         "ChanObjectType: peer\r\n" /* "peer" or "user" */
         "IPaddress: %s\r\n"
         "IPport: %d\r\n"
         "Dynamic: %s\r\n"
         "Natsupport: %s\r\n"
         "VideoSupport: %s\r\n"
         "TextSupport: %s\r\n"
         "ACL: %s\r\n"
         "Status: %s\r\n"
         "RealtimeDevice: %s\r\n\r\n", 
         idtext,
         peer->name, 
         peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "-none-",
         ntohs(peer->addr.sin_port), 
         peer->host_dynamic ? "yes" : "no",  /* Dynamic or not? */
         ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE) ? "yes" : "no",  /* NAT=yes? */
         ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no",  /* VIDEOSUPPORT=yes? */
         ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "yes" : "no",   /* TEXTSUPPORT=yes? */
         peer->ha ? "yes" : "no",       /* permit/deny */
         status,
         realtimepeers ? (peer->is_realtime ? "yes":"no") : "no");
      }
      ao2_unlock(peer);
      peer = peerarray[k] = unref_peer(peer, "toss iterator peer ptr");
   }
   
   if (!s)
      ast_cli(fd, "%d sip peers [Monitored: %d online, %d offline Unmonitored: %d online, %d offline]\n",
              total_peers, peers_mon_online, peers_mon_offline, peers_unmon_online, peers_unmon_offline);

   if (havepattern)
      regfree(&regexbuf);

   if (total)
      *total = total_peers;
   
   ast_free(peerarray);
   
   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}

static void * _sip_tcp_helper_thread	(	struct sip_pvt *	pvt,
		struct ast_tcptls_session_instance *	tcptls_session
	)			[static]

SIP TCP thread management function This function reads from the socket, parses the packet into a request.

Todo:

XXX If there's no Content-Length or if the content-length and what we receive is not the same - we should generate an error

Definition at line 2955 of file chan_sip.c.

References sip_threadinfo::alert_pipe, ao2_lock(), ao2_ref, ao2_t_find, ao2_t_ref, ao2_t_unlink, ao2_unlock(), ast_atomic_fetchadd_int(), ast_debug, AST_LIST_REMOVE_HEAD, ast_log(), ast_mutex_lock(), ast_mutex_unlock(), ast_poll, ast_str_append(), ast_str_buffer(), ast_str_create(), ast_str_reset(), ast_tcptls_client_start(), ast_tcptls_server_write(), ast_wait_for_input(), sip_request::authenticated, buf, cleanup(), ast_tcptls_session_instance::client, copy_request(), tcptls_packet::data, sip_request::data, deinit_req(), errno, ast_tcptls_session_instance::f, sip_socket::fd, ast_tcptls_session_instance::fd, get_header(), handle_request_do(), tcptls_packet::len, sip_request::len, ast_tcptls_session_instance::lock, LOG_ERROR, LOG_WARNING, me, MIN, OBJ_POINTER, sip_threadinfo::packet_q, ast_tcptls_session_instance::parent, parse_request(), sip_socket::port, ast_tcptls_session_instance::remote_address, REQ_OFFSET_TO_STR, set_socket_transport(), sip_check_authtimeout(), SIP_MIN_PACKET, sip_threadinfo_create(), SIP_TRANSPORT_TCP, SIP_TRANSPORT_TLS, sip_request::socket, ast_tcptls_session_instance::ssl, sip_threadinfo::stop, TCPTLS_ALERT_DATA, TCPTLS_ALERT_STOP, sip_socket::tcptls_session, sip_threadinfo::tcptls_session, and sip_threadinfo::threadid.

Referenced by sip_tcp_worker_fn().

{
   int res, cl, timeout = -1, authenticated = 0, flags, after_poll = 0, need_poll = 1;
   time_t start;
   struct sip_request req = { 0, } , reqcpy = { 0, };
   struct sip_threadinfo *me = NULL;
   char buf[1024] = "";
   struct pollfd fds[2] = { { 0 }, { 0 }, };
   struct ast_tcptls_session_args *ca = NULL;

   /* If this is a server session, then the connection has already been
    * setup. Check if the authlimit has been reached and if not create the
    * threadinfo object so we can access this thread for writing.
    *
    * if this is a client connection more work must be done.
    * 1. We own the parent session args for a client connection.  This pointer needs
    *    to be held on to so we can decrement it's ref count on thread destruction.
    * 2. The threadinfo object was created before this thread was launched, however
    *    it must be found within the threadt table.
    * 3. Last, the tcptls_session must be started.
    */
   if (!tcptls_session->client) {
      if (ast_atomic_fetchadd_int(&unauth_sessions, +1) >= authlimit) {
         /* unauth_sessions is decremented in the cleanup code */
         goto cleanup;
      }

      if ((flags = fcntl(tcptls_session->fd, F_GETFL)) == -1) {
         ast_log(LOG_ERROR, "error setting socket to non blocking mode, fcntl() failed: %s\n", strerror(errno));
         goto cleanup;
      }

      flags |= O_NONBLOCK;
      if (fcntl(tcptls_session->fd, F_SETFL, flags) == -1) {
         ast_log(LOG_ERROR, "error setting socket to non blocking mode, fcntl() failed: %s\n", strerror(errno));
         goto cleanup;
      }

      if (!(me = sip_threadinfo_create(tcptls_session, tcptls_session->ssl ? SIP_TRANSPORT_TLS : SIP_TRANSPORT_TCP))) {
         goto cleanup;
      }
      ao2_t_ref(me, +1, "Adding threadinfo ref for tcp_helper_thread");
   } else {
      struct sip_threadinfo tmp = {
         .tcptls_session = tcptls_session,
      };

      if ((!(ca = tcptls_session->parent)) ||
         (!(me = ao2_t_find(threadt, &tmp, OBJ_POINTER, "ao2_find, getting sip_threadinfo in tcp helper thread"))) ||
         (!(tcptls_session = ast_tcptls_client_start(tcptls_session)))) {
         goto cleanup;
      }
   }

   flags = 1;
   if (setsockopt(tcptls_session->fd, SOL_SOCKET, SO_KEEPALIVE, &flags, sizeof(flags))) {
      ast_log(LOG_ERROR, "error enabling TCP keep-alives on sip socket: %s\n", strerror(errno));
      goto cleanup;
   }

   me->threadid = pthread_self();
   ast_debug(2, "Starting thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");

   /* set up pollfd to watch for reads on both the socket and the alert_pipe */
   fds[0].fd = tcptls_session->fd;
   fds[1].fd = me->alert_pipe[0];
   fds[0].events = fds[1].events = POLLIN | POLLPRI;

   if (!(req.data = ast_str_create(SIP_MIN_PACKET)))
      goto cleanup;
   if (!(reqcpy.data = ast_str_create(SIP_MIN_PACKET)))
      goto cleanup;

   if(time(&start) == -1) {
      ast_log(LOG_ERROR, "error executing time(): %s\n", strerror(errno));
      goto cleanup;
   }

   for (;;) {
      struct ast_str *str_save;

      if (!tcptls_session->client && req.authenticated && !authenticated) {
         authenticated = 1;
         ast_atomic_fetchadd_int(&unauth_sessions, -1);
      }

      /* calculate the timeout for unauthenticated server sessions */
      if (!tcptls_session->client && !authenticated ) {
         if ((timeout = sip_check_authtimeout(start)) < 0) {
            goto cleanup;
         }

         if (timeout == 0) {
            ast_debug(2, "SIP %s server timed out\n", tcptls_session->ssl ? "SSL": "TCP");
            goto cleanup;
         }
      } else {
         timeout = -1;
      }

      res = ast_poll(fds, 2, timeout); /* polls for both socket and alert_pipe */
      if (res < 0) {
         ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", tcptls_session->ssl ? "SSL": "TCP", res);
         goto cleanup;
      } else if (res == 0) {
         /* timeout */
         ast_debug(2, "SIP %s server timed out\n", tcptls_session->ssl ? "SSL": "TCP");
         goto cleanup;
      }

      /* handle the socket event, check for both reads from the socket fd,
       * and writes from alert_pipe fd */
      if (fds[0].revents) { /* there is data on the socket to be read */
         after_poll = 1;

         fds[0].revents = 0;

         /* clear request structure */
         str_save = req.data;
         memset(&req, 0, sizeof(req));
         req.data = str_save;
         ast_str_reset(req.data);

         str_save = reqcpy.data;
         memset(&reqcpy, 0, sizeof(reqcpy));
         reqcpy.data = str_save;
         ast_str_reset(reqcpy.data);

         memset(buf, 0, sizeof(buf));

         if (tcptls_session->ssl) {
            set_socket_transport(&req.socket, SIP_TRANSPORT_TLS);
            req.socket.port = htons(ourport_tls);
         } else {
            set_socket_transport(&req.socket, SIP_TRANSPORT_TCP);
            req.socket.port = htons(ourport_tcp);
         }
         req.socket.fd = tcptls_session->fd;

         /* Read in headers one line at a time */
         while (req.len < 4 || strncmp(REQ_OFFSET_TO_STR(&req, len - 4), "\r\n\r\n", 4)) {
            if (!tcptls_session->client && !authenticated ) {
               if ((timeout = sip_check_authtimeout(start)) < 0) {
                  goto cleanup;
               }

               if (timeout == 0) {
                  ast_debug(2, "SIP %s server timed out\n", tcptls_session->ssl ? "SSL": "TCP");
                  goto cleanup;
               }
            } else {
               timeout = -1;
            }

            if (!tcptls_session->ssl || need_poll) {
               need_poll = 0;
               after_poll = 1;
               res = ast_wait_for_input(tcptls_session->fd, timeout);
               if (res < 0) {
                  ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", tcptls_session->ssl ? "SSL": "TCP", res);
                  goto cleanup;
               } else if (res == 0) {
                  /* timeout */
                  ast_debug(2, "SIP %s server timed out\n", tcptls_session->ssl ? "SSL": "TCP");
                  goto cleanup;
               }
            }

            ast_mutex_lock(&tcptls_session->lock);
            if (!fgets(buf, sizeof(buf), tcptls_session->f)) {
               ast_mutex_unlock(&tcptls_session->lock);
               if (after_poll) {
                  goto cleanup;
               } else {
                  need_poll = 1;
                  continue;
               }
            }
            ast_mutex_unlock(&tcptls_session->lock);
            after_poll = 0;
            if (me->stop)
                goto cleanup;
            ast_str_append(&req.data, 0, "%s", buf);
            req.len = req.data->used;
         }
         copy_request(&reqcpy, &req);
         parse_request(&reqcpy);
         /* In order to know how much to read, we need the content-length header */
         if (sscanf(get_header(&reqcpy, "Content-Length"), "%30d", &cl)) {
            while (cl > 0) {
               size_t bytes_read;
               if (!tcptls_session->client && !authenticated ) {
                  if ((timeout = sip_check_authtimeout(start)) < 0) {
                     goto cleanup;
                  }

                  if (timeout == 0) {
                     ast_debug(2, "SIP %s server timed out\n", tcptls_session->ssl ? "SSL": "TCP");
                     goto cleanup;
                  }
               } else {
                  timeout = -1;
               }

               if (!tcptls_session->ssl || need_poll) {
                  res = ast_wait_for_input(tcptls_session->fd, timeout);
                  if (res < 0) {
                     ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", tcptls_session->ssl ? "SSL": "TCP", res);
                     goto cleanup;
                  } else if (res == 0) {
                     /* timeout */
                     ast_debug(2, "SIP %s server timed out\n", tcptls_session->ssl ? "SSL": "TCP");
                     goto cleanup;
                  }
               }

               ast_mutex_lock(&tcptls_session->lock);
               if (!(bytes_read = fread(buf, 1, MIN(sizeof(buf) - 1, cl), tcptls_session->f))) {
                  ast_mutex_unlock(&tcptls_session->lock);
                  if (after_poll) {
                     goto cleanup;
                  } else {
                     need_poll = 1;
                     continue;
                  }
               }
               buf[bytes_read] = '\0';
               ast_mutex_unlock(&tcptls_session->lock);
               after_poll = 0;
               if (me->stop)
                  goto cleanup;
               cl -= strlen(buf);
               ast_str_append(&req.data, 0, "%s", buf);
               req.len = req.data->used;
            }
         }
         /*! \todo XXX If there's no Content-Length or if the content-length and what
               we receive is not the same - we should generate an error */

         req.socket.tcptls_session = tcptls_session;
         handle_request_do(&req, &tcptls_session->remote_address);
      }

      if (fds[1].revents) { /* alert_pipe indicates there is data in the send queue to be sent */
         enum sip_tcptls_alert alert;
         struct tcptls_packet *packet;

         fds[1].revents = 0;

         if (read(me->alert_pipe[0], &alert, sizeof(alert)) == -1) {
            ast_log(LOG_ERROR, "read() failed: %s\n", strerror(errno));
            continue;
         }

         switch (alert) {
         case TCPTLS_ALERT_STOP:
            goto cleanup;
         case TCPTLS_ALERT_DATA:
            ao2_lock(me);
            if (!(packet = AST_LIST_REMOVE_HEAD(&me->packet_q, entry))) {
               ast_log(LOG_WARNING, "TCPTLS thread alert_pipe indicated packet should be sent, but frame_q is empty");
            }
            ao2_unlock(me);

            if (packet) {
               if (ast_tcptls_server_write(tcptls_session, ast_str_buffer(packet->data), packet->len) == -1) {
                  ast_log(LOG_WARNING, "Failure to write to tcp/tls socket\n");
               }
               ao2_t_ref(packet, -1, "tcptls packet sent, this is no longer needed");
            }
            break;
         default:
            ast_log(LOG_ERROR, "Unknown tcptls thread alert '%d'\n", alert);
         }
      }
   }

   ast_debug(2, "Shutting down thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");

cleanup:
   if (tcptls_session && !tcptls_session->client && !authenticated) {
      ast_atomic_fetchadd_int(&unauth_sessions, -1);
   }

   if (me) {
      ao2_t_unlink(threadt, me, "Removing tcptls helper thread, thread is closing");
      ao2_t_ref(me, -1, "Removing tcp_helper_threads threadinfo ref");
   }
   deinit_req(&reqcpy);
   deinit_req(&req);

   /* if client, we own the parent session arguments and must decrement ref */
   if (ca) {
      ao2_t_ref(ca, -1, "closing tcptls thread, getting rid of client tcptls_session arguments");
   }

   if (tcptls_session) {
      ast_mutex_lock(&tcptls_session->lock);
      if (tcptls_session->f) {
         fclose(tcptls_session->f);
         tcptls_session->f = NULL;
      }
      if (tcptls_session->fd != -1) {
         close(tcptls_session->fd);
         tcptls_session->fd = -1;
      }
      tcptls_session->parent = NULL;
      ast_mutex_unlock(&tcptls_session->lock);

      ao2_ref(tcptls_session, -1);
      tcptls_session = NULL;
   }
   return NULL;
}

static int acf_channel_read	(	struct ast_channel *	chan,
		const char *	funcname,
		char *	preparse,
		char *	buf,
		size_t	buflen
	)			[static]

Definition at line 21669 of file chan_sip.c.

References AST_APP_ARG, ast_copy_string(), AST_DECLARE_APP_ARGS, ast_inet_ntoa(), ast_log(), ast_rtp_get_peer(), ast_rtp_get_qos(), ast_rtp_get_quality(), AST_STANDARD_APP_ARGS, ast_strdupa, ast_strlen_zero(), sip_pvt::from, IS_SIP_TECH, ast_rtp_quality::local_count, ast_rtp_quality::local_jitter, ast_rtp_quality::local_lostpackets, ast_rtp_quality::local_ssrc, LOG_ERROR, LOG_WARNING, parse(), sip_pvt::peername, sip_pvt::recv, ast_rtp_quality::remote_count, ast_rtp_quality::remote_jitter, ast_rtp_quality::remote_lostpackets, ast_rtp_quality::remote_ssrc, sip_pvt::rtp, RTPQOS_SUMMARY, ast_rtp_quality::rtt, sip_pvt::sa, t38properties::state, sip_pvt::t38, T38_DISABLED, ast_channel::tech, ast_channel::tech_pvt, sip_pvt::trtp, sip_pvt::uri, sip_pvt::useragent, and sip_pvt::vrtp.

{
   struct sip_pvt *p = chan->tech_pvt;
   char *all = "", *parse = ast_strdupa(preparse);
   int res = 0;
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(param);
      AST_APP_ARG(type);
      AST_APP_ARG(field);
   );
   AST_STANDARD_APP_ARGS(args, parse);

   /* Sanity check */
   if (!IS_SIP_TECH(chan->tech)) {
      ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
      return 0;
   }

   memset(buf, 0, buflen);

   if (p == NULL) {
      return -1;
   }

   if (!strcasecmp(args.param, "peerip")) {
      ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(p->sa.sin_addr) : "", buflen);
   } else if (!strcasecmp(args.param, "recvip")) {
      ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(p->recv.sin_addr) : "", buflen);
   } else if (!strcasecmp(args.param, "from")) {
      ast_copy_string(buf, p->from, buflen);
   } else if (!strcasecmp(args.param, "uri")) {
      ast_copy_string(buf, p->uri, buflen);
   } else if (!strcasecmp(args.param, "useragent")) {
      ast_copy_string(buf, p->useragent, buflen);
   } else if (!strcasecmp(args.param, "peername")) {
      ast_copy_string(buf, p->peername, buflen);
   } else if (!strcasecmp(args.param, "t38passthrough")) {
      ast_copy_string(buf, (p->t38.state == T38_DISABLED) ? "0" : "1", buflen);
   } else if (!strcasecmp(args.param, "rtpdest")) {
      struct sockaddr_in sin;
      struct ast_rtp *stream = NULL;

      if (ast_strlen_zero(args.type))
         args.type = "audio";

      if (!strcasecmp(args.type, "audio")) {
         stream = p->rtp;
      } else if (!strcasecmp(args.type, "video")) {
         stream = p->vrtp;
      } else if (!strcasecmp(args.type, "text")) {
         stream = p->trtp;
      } else {
         return -1;
      }

      if (!stream) {
         return -1;
      }

      ast_rtp_get_peer(stream, &sin);
      snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
   } else if (!strcasecmp(args.param, "rtpqos")) {
      struct ast_rtp_quality qos;
      struct ast_rtp *rtp = p->rtp;
      
      memset(&qos, 0, sizeof(qos));

      if (ast_strlen_zero(args.type))
         args.type = "audio";
      if (ast_strlen_zero(args.field))
         args.field = "all";
      
      if (!strcasecmp(args.type, "AUDIO")) {
         all = ast_rtp_get_quality(rtp = p->rtp, &qos, RTPQOS_SUMMARY);
      } else if (!strcasecmp(args.type, "VIDEO")) {
         all = ast_rtp_get_quality(rtp = p->vrtp, &qos, RTPQOS_SUMMARY);
      } else if (!strcasecmp(args.type, "TEXT")) {
         all = ast_rtp_get_quality(rtp = p->trtp, &qos, RTPQOS_SUMMARY);
      } else {
         return -1;
      }
      
      if (!strcasecmp(args.field, "local_ssrc"))
         snprintf(buf, buflen, "%u", qos.local_ssrc);
      else if (!strcasecmp(args.field, "local_lostpackets"))
         snprintf(buf, buflen, "%u", qos.local_lostpackets);
      else if (!strcasecmp(args.field, "local_jitter"))
         snprintf(buf, buflen, "%.0f", qos.local_jitter * 1000.0);
      else if (!strcasecmp(args.field, "local_count"))
         snprintf(buf, buflen, "%u", qos.local_count);
      else if (!strcasecmp(args.field, "remote_ssrc"))
         snprintf(buf, buflen, "%u", qos.remote_ssrc);
      else if (!strcasecmp(args.field, "remote_lostpackets"))
         snprintf(buf, buflen, "%u", qos.remote_lostpackets);
      else if (!strcasecmp(args.field, "remote_jitter"))
         snprintf(buf, buflen, "%.0f", qos.remote_jitter * 1000.0);
      else if (!strcasecmp(args.field, "remote_count"))
         snprintf(buf, buflen, "%u", qos.remote_count);
      else if (!strcasecmp(args.field, "rtt"))
         snprintf(buf, buflen, "%.0f", qos.rtt * 1000.0);
      else if (!strcasecmp(args.field, "all"))
         ast_copy_string(buf, all, buflen);
      else if (!ast_rtp_get_qos(rtp, args.field, buf, buflen))
          ;
      else {
         ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
         return -1;
      }
   } else {
      res = -1;
   }
   return res;
}

static void add_blank

(

struct sip_request *

req

)

[static]

add a blank line if no body

Definition at line 4425 of file chan_sip.c.

References ast_str_append(), ast_str_strlen(), sip_request::data, sip_request::len, and sip_request::lines.

Referenced by send_request(), and send_response().

{
   if (!req->lines) {
      /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
      ast_str_append(&req->data, 0, "\r\n");
      req->len = ast_str_strlen(req->data);
   }
}

static void add_codec_to_sdp	(	const struct sip_pvt *	p,
		int	codec,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug,
		int *	min_packet_size
	)			[static]

Add codec offer to SDP offer/answer body in INVITE or 200 OK.

Definition at line 10408 of file chan_sip.c.

References ast_codec_pref_getsize(), AST_FORMAT_G723_1, AST_FORMAT_G729A, AST_FORMAT_ILBC, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, ast_getformatname(), ast_rtp_codec_getpref(), ast_rtp_lookup_code(), ast_rtp_lookup_mime_subtype(), ast_rtp_lookup_sample_rate(), AST_RTP_OPT_G726_NONSTANDARD, ast_str_append(), ast_test_flag, ast_verbose, ast_format_list::cur_ms, sip_pvt::flags, sip_pvt::rtp, and SIP_G726_NONSTANDARD.

Referenced by add_sdp().

{
   int rtp_code;
   struct ast_format_list fmt;


   if (debug)
      ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
   if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1)
      return;

   if (p->rtp) {
      struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
      fmt = ast_codec_pref_getsize(pref, codec);
   } else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
      return;
   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
             ast_rtp_lookup_mime_subtype(1, codec,
                     ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
             ast_rtp_lookup_sample_rate(1, codec));

   switch (codec) {
   case AST_FORMAT_G729A:
      /* Indicate that we don't support VAD (G.729 annex B) */
      ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code);
      break;
   case AST_FORMAT_G723_1:
      /* Indicate that we don't support VAD (G.723.1 annex A) */
      ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
      break;
   case AST_FORMAT_ILBC:
      /* Add information about us using only 20/30 ms packetization */
      ast_str_append(a_buf, 0, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
      break;
   case AST_FORMAT_SIREN7:
      /* Indicate that we only expect 32Kbps */
      ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=32000\r\n", rtp_code);
      break;
   case AST_FORMAT_SIREN14:
      /* Indicate that we only expect 48Kbps */
      ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code);
      break;
   }

   if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
      *min_packet_size = fmt.cur_ms;

   /* Our first codec packetization processed cannot be zero */
   if ((*min_packet_size)==0 && fmt.cur_ms)
      *min_packet_size = fmt.cur_ms;
}

static int add_content	(	struct sip_request *	req,
		const char *	line
	)			[static]

Add content (not header) to SIP message.

Definition at line 9609 of file chan_sip.c.

References ast_log(), ast_str_append(), sip_request::content, sip_request::lines, and LOG_WARNING.

Referenced by add_digit(), add_sdp(), add_text(), add_vidupdate(), transmit_notify_with_mwi(), transmit_notify_with_sipfrag(), and transmit_state_notify().

{
   if (req->lines) {
      ast_log(LOG_WARNING, "Can't add more content when the content has been finalized\n");
      return -1;
   }

   ast_str_append(&req->content, 0, "%s", line);
   return 0;
}

static int add_digit	(	struct sip_request *	req,
		char	digit,
		unsigned int	duration,
		int	mode
	)			[static]

Add DTMF INFO tone to sip message Mode = 0 for application/dtmf-relay (Cisco) 1 for application/dtmf.

Definition at line 10362 of file chan_sip.c.

References add_content(), and add_header().

Referenced by transmit_info_with_digit().

{
   char tmp[256];
   int event;
   if (mode) {
      /* Application/dtmf short version used by some implementations */
      if (digit == '*')
         event = 10;
      else if (digit == '#')
         event = 11;
      else if ((digit >= 'A') && (digit <= 'D'))
         event = 12 + digit - 'A';
      else
         event = atoi(&digit);
      snprintf(tmp, sizeof(tmp), "%d\r\n", event);
      add_header(req, "Content-Type", "application/dtmf");
      add_content(req, tmp);
   } else {
      /* Application/dtmf-relay as documented by Cisco */
      snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=%u\r\n", digit, duration);
      add_header(req, "Content-Type", "application/dtmf-relay");
      add_content(req, tmp);
   }
   return 0;
}

static int add_header	(	struct sip_request *	req,
		const char *	var,
		const char *	value
	)			[static]

Add header to SIP message.

Definition at line 9562 of file chan_sip.c.

References ast_log(), ast_str_append(), ast_str_strlen(), sip_settings::compactheaders, sip_request::data, find_alias(), sip_request::header, sip_request::headers, sip_request::len, sip_request::lines, LOG_WARNING, sip_cfg, and SIP_MAX_HEADERS.

{
   if (req->headers == SIP_MAX_HEADERS) {
      ast_log(LOG_WARNING, "Out of SIP header space\n");
      return -1;
   }

   if (req->lines) {
      ast_log(LOG_WARNING, "Can't add more headers when lines have been added\n");
      return -1;
   }

   if (sip_cfg.compactheaders) {
      var = find_alias(var, var);
   }

   ast_str_append(&req->data, 0, "%s: %s\r\n", var, value);
   req->header[req->headers] = req->len;

   req->len = ast_str_strlen(req->data);
   req->headers++;

   return 0;   
}

static void add_noncodec_to_sdp	(	const struct sip_pvt *	p,
		int	format,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug
	)			[static]

Add RFC 2833 DTMF offer to SDP.

Definition at line 10541 of file chan_sip.c.

References AST_RTP_DTMF, ast_rtp_lookup_code(), ast_rtp_lookup_mime_subtype(), ast_rtp_lookup_sample_rate(), ast_str_append(), ast_verbose, and sip_pvt::rtp.

Referenced by add_sdp().

{
   int rtp_code;

   if (debug)
      ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype(0, format, 0));
   if ((rtp_code = ast_rtp_lookup_code(p->rtp, 0, format)) == -1)
      return;

   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
             ast_rtp_lookup_mime_subtype(0,