@ -282,11 +282,16 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
# define TRUE 1
# endif
# define SIPBUFSIZE 512
# ifndef MAX
# define MAX(a,b) ((a) > (b) ? (a) : (b))
# endif
# define SIPBUFSIZE 512 /*!< Buffer size for many operations */
# define XMIT_ERROR -2
# define SIP_RESERVED "; / ?:@&=+$,# "
# define SIP_RESERVED "; / ?:@&=+$,# " /*!< Reserved characters in the username part of the URI */
/* #define VOCAL_DATA_HACK */
@ -315,10 +320,6 @@ static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registr
static int default_expiry = DEFAULT_DEFAULT_EXPIRY ;
static int mwi_expiry = DEFAULT_MWI_EXPIRY ;
# ifndef MAX
# define MAX(a,b) ((a) > (b) ? (a) : (b))
# endif
# define CALLERID_UNKNOWN "Unknown"
# define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
@ -327,19 +328,18 @@ static int mwi_expiry = DEFAULT_MWI_EXPIRY;
# define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
# define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
# define SIP_TIMER_T1 500 /* SIP timer T1 (according to RFC 3261) */
# define SIP_TIMER_T1 500 /* !< SIP timer T1 (according to RFC 3261) */
# define SIP_TRANS_TIMEOUT 64 * SIP_TIMER_T1 / *!< SIP request timeout (rfc 3261) 64*T1
\ todo Use known T1 for timeout ( peerpoke )
*/
# define DEFAULT_TRANS_TIMEOUT -1 /* Use default SIP transaction timeout */
# define DEFAULT_TRANS_TIMEOUT -1 /* !< Use default SIP transaction timeout */
# define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
# define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
# define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
# define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */
# define SIP_MIN_PACKET 1024 /*!< Initialize size of memory to allocate for packets */
# define INITIAL_CSEQ 101 /*!< o ur initial sip sequence number */
# define INITIAL_CSEQ 101 /*!< O ur initial sip sequence number */
# define DEFAULT_MAX_SE 1800 /*!< Session-Timer Default Session-Expires period (RFC 4028) */
# define DEFAULT_MIN_SE 90 /*!< Session-Timer Default Min-SE period (RFC 4028) */
@ -354,15 +354,16 @@ static struct ast_jb_conf default_jbconf =
. resync_threshold = - 1 ,
. impl = " "
} ;
static struct ast_jb_conf global_jbconf ; /*!< Global jitterbuffer configuration */
static struct ast_jb_conf global_jbconf ; /*!< Global jitterbuffer configuration */
static const char config [ ] = " sip.conf " ; /*!< Main configuration file */
static const char config [ ] = " sip.conf " ; /*!< Main configuration file */
static const char notify_config [ ] = " sip_notify.conf " ; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
# define RTP 1
# define NO_RTP 0
/*! \brief Authorization scheme for call transfers
\ note Not a bitfield flag , since there are plans for other modes ,
like " only allow transfers for authenticated devices " */
enum transfermodes {
@ -373,8 +374,8 @@ enum transfermodes {
/*! \brief The result of a lot of functions */
enum sip_result {
AST_SUCCESS = 0 , /*! FALSE means success, funny enough */
AST_FAILURE = - 1 ,
AST_SUCCESS = 0 , /*! < FALSE means success, funny enough */
AST_FAILURE = - 1 , /*!< Failure code */
} ;
/*! \brief States for the INVITE transaction, not the dialog
@ -418,13 +419,14 @@ enum xmittype {
XMIT_UNRELIABLE = 0 , /*!< Transmit SIP message without bothering with re-transmits */
} ;
/*! \brief Results from the parse_register() function */
enum parse_register_result {
PARSE_REGISTER_FAILED ,
PARSE_REGISTER_UPDATE ,
PARSE_REGISTER_QUERY ,
} ;
/*! \brief Type of subscription, based on the packages we do support */
/*! \brief Type of subscription, based on the packages we do support , see \ref subscription_types */
enum subscriptiontype {
NONE = 0 ,
XPIDF_XML ,
@ -757,16 +759,16 @@ static const struct cfsip_options {
yet encouraging new behaviour on new installations
*/
/*@{*/
# define DEFAULT_CONTEXT "default"
# define DEFAULT_MOHINTERPRET "default"
# define DEFAULT_CONTEXT "default" /*!< The default context for [general] section as well as devices */
# define DEFAULT_MOHINTERPRET "default" /*!< The default music class */
# define DEFAULT_MOHSUGGEST ""
# define DEFAULT_VMEXTEN "asterisk"
# define DEFAULT_CALLERID "asterisk"
# define DEFAULT_VMEXTEN "asterisk" /*!< Default voicemail extension */
# define DEFAULT_CALLERID "asterisk" /*!< Default caller ID */
# define DEFAULT_NOTIFYMIME "application / simple-message-summary"
# define DEFAULT_ALLOWGUEST TRUE
# define DEFAULT_CALLCOUNTER FALSE
# define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */
# define DEFAULT_COMPACTHEADERS FALSE
# define DEFAULT_COMPACTHEADERS FALSE /*!< Send compact (one-character) SIP headers. Default off */
# define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
# define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
# define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
@ -777,10 +779,10 @@ static const struct cfsip_options {
# define DEFAULT_COS_TEXT 5 /*!< Level 2 class of service for text media (T.140) */
# define DEFAULT_ALLOW_EXT_DOM TRUE /*!< Allow external domains */
# define DEFAULT_REALM "asterisk" /*!< Realm for HTTP digest authentication */
# define DEFAULT_NOTIFYRINGING TRUE
# define DEFAULT_PEDANTIC FALSE
# define DEFAULT_AUTOCREATEPEER FALSE
# define DEFAULT_QUALIFY FALSE
# define DEFAULT_NOTIFYRINGING TRUE /*!< Notify devicestate system on ringing state */
# define DEFAULT_PEDANTIC FALSE /*!< Avoid following SIP standards for dialog matching */
# define DEFAULT_AUTOCREATEPEER FALSE /*!< Don't create peers automagically */
# define DEFAULT_QUALIFY FALSE /*!< Don't monitor devices */
# define DEFAULT_REGEXTENONQUALIFY FALSE
# define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
# define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
@ -4233,13 +4235,9 @@ static struct sip_peer *realtime_peer(const char *newpeername, struct sockaddr_i
* This is used on find matching device on name or ip / port .
If the device was declared as type = peer , we don ' t match on peer name on incoming INVITEs .
\ note Avoid using this function in new functions if there is a way to avoid it , i
\ note Avoid using this function in new functions if there is a way to avoid it ,
since it might cause a database lookup .
\ todo - we need to fix so that we actually match on username only if forcenamematch is on .
There ' s a flag in peers for " onlymatchonip " - these peers needs to be avoided when
searching the " peers " hash table .
*/
static struct sip_peer * find_peer ( const char * peer , struct sockaddr_in * sin , int realtime , int forcenamematch , int devstate_only )
{
@ -4247,6 +4245,7 @@ static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int
struct sip_peer tmp_peer ;
auto int find_by_name ( void * obj , void * arg , int flags ) ;
int find_by_name ( void * obj , void * arg , int flags )
{
struct sip_peer * search = obj , * match = arg ;
@ -4279,8 +4278,9 @@ static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int
}
}
if ( ! p & & ( realtime | | devstate_only ) )
if ( ! p & & ( realtime | | devstate_only ) ) {
p = realtime_peer ( peer , sin , devstate_only ) ;
}
return p ;
}
Olle Johansson
18 years ago