@ -1152,13 +1152,13 @@ static const char *sip_reason_code_to_str(enum AST_REDIRECTING_REASON code)
configuring devices
*/
/*@{*/
static char default_language [ MAX_LANGUAGE ] ; /*! Default language setting for new channels */
static char default_callerid [ AST_MAX_EXTENSION ] ;
static char default_mwi_from [ 80 ] ;
static char default_fromdomain [ AST_MAX_EXTENSION ] ;
static char default_notifymime [ AST_MAX_EXTENSION ] ;
static char default_language [ MAX_LANGUAGE ] ; /*!< Default language setting for new channels */
static char default_callerid [ AST_MAX_EXTENSION ] ; /*!< Default caller ID for sip messages */
static char default_mwi_from [ 80 ] ; /*!< Default caller ID for MWI updates */
static char default_fromdomain [ AST_MAX_EXTENSION ] ; /*!< Default domain on outound messages */
static char default_notifymime [ AST_MAX_EXTENSION ] ; /*!< Default MIME media type for MWI notify messages */
static char default_vmexten [ AST_MAX_EXTENSION ] ; /*!< Default From Username on MWI updates */
static int default_qualify ; /*!< Default Qualify= setting */
static char default_vmexten [ AST_MAX_EXTENSION ] ;
static char default_mohinterpret [ MAX_MUSICCLASS ] ; /*!< Global setting for moh class to use when put on hold */
static char default_mohsuggest [ MAX_MUSICCLASS ] ; /*!< Global setting for moh class to suggest when putting
* a bridged channel on hold */
@ -1177,6 +1177,7 @@ static unsigned int default_primary_transport; /*!< Default primary Transport (
*/
/*@{*/
/*! \brief a place to store all global settings for the sip channel driver
These are settings that will be possibly to apply on a group level later on .
\ note Do not add settings that only apply to the channel itself and can ' t
be applied to devices ( trunks , services , phones )
@ -1342,10 +1343,8 @@ struct sip_request {
char debug ; /*!< print extra debugging if non zero */
char has_to_tag ; /*!< non-zero if packet has To: tag */
char ignore ; /*!< if non-zero This is a re-transmit, ignore it */
/* Array of offsets into the request string of each SIP header*/
ptrdiff_t header [ SIP_MAX_HEADERS ] ;
/* Array of offsets into the request string of each SDP line*/
ptrdiff_t line [ SIP_MAX_LINES ] ;
ptrdiff_t header [ SIP_MAX_HEADERS ] ; /*!< Array of offsets into the request string of each SIP header*/
ptrdiff_t line [ SIP_MAX_LINES ] ; /*!< Array of offsets into the request string of each SDP line*/
struct ast_str * data ;
/* XXX Do we need to unref socket.ser when the request goes away? */
struct sip_socket socket ; /*!< The socket used for this request */
@ -1845,10 +1844,11 @@ struct sip_pvt {
* By doing this , even if we don ' t want to answer a particular media stream with something meaningful , we can
* still put an m = line in our answer with the port set to 0.
*
* The reason for the length being 4 is that in this branch of Asterisk , the only media types supported are
* The reason for the length being 4 ( OFFERED_MEDIA_COUNT ) is that in this branch of Asterisk , the only media types supported are
* image , audio , text , and video . Therefore we need to keep track of which types of media were offered .
* Note that secure RTP defines new types of SDP media .
*
* Note that i f we wanted to be 100 % correct , we would keep a list of all media streams offered . That way we could respond
* I f we wanted to be 100 % correct , we would keep a list of all media streams offered . That way we could respond
* even to unknown media types , and we could respond to multiple streams of the same type . Such large - scale changes
* are not a good idea for released branches , though , so we ' re compromising by just making sure that for the common cases :
* audio and video , audio and T .38 , and audio and text , we give the appropriate response to both media streams .
@ -2016,7 +2016,7 @@ struct sip_peer {
/*! Mailboxes that this peer cares about */
AST_LIST_HEAD_NOLOCK ( , sip_mailbox ) mailboxes ;
int maxcallbitrate ; /*!< Maximum Bitrate for a video call */
int maxcallbitrate ; /*!< Maximum Bitrate for a video call */
int expire ; /*!< When to expire this peer registration */
int capability ; /*!< Codec capability */
int rtptimeout ; /*!< RTP timeout */
@ -2027,13 +2027,12 @@ struct sip_peer {
struct sip_proxy * outboundproxy ; /*!< Outbound proxy for this peer */
struct ast_dnsmgr_entry * dnsmgr ; /*!< DNS refresh manager for peer */
struct sockaddr_in addr ; /*!< IP address of peer */
/* Qualification */
struct sip_pvt * call ; /*!< Call pointer */
int pokeexpire ; /*!< When to expire poke (qualify= checking) */
int lastms ; /*!< How long last response took (in ms), or -1 for no response */
int maxms ; /*!< Max ms we will accept for the host to be up, 0 to not monitor */
int qualifyfreq ; /*!< Qualification: How often to check for the host to be up */
struct timeval ps ; /*!< Time for sending SIP OPTION in sip_pke_peer() */
int pokeexpire ; /*!< Qualification: When to expire poke (qualify= checking) */
int lastms ; /*!< Qualification: How long last response took (in ms), or -1 for no response */
int maxms ; /*!< Qualification: Max ms we will accept for the host to be up, 0 to not monitor */
int qualifyfreq ; /*!< Qualification: Qualification: How often to check for the host to be up */
struct timeval ps ; /*!< Qualification: Time for sending SIP OPTION in sip_pke_peer() */
struct sockaddr_in defaddr ; /*!< Default IP address, used until registration */
struct ast_ha * ha ; /*!< Access control list */
struct ast_ha * contactha ; /*!< Restrict what IPs are allowed in the Contact header (for registration) */
@ -2078,7 +2077,6 @@ struct sip_registry {
AST_STRING_FIELD ( secret ) ; /*!< Password in clear text */
AST_STRING_FIELD ( md5secret ) ; /*!< Password in md5 */
AST_STRING_FIELD ( callback ) ; /*!< Contact extension */
AST_STRING_FIELD ( random ) ;
AST_STRING_FIELD ( peername ) ; /*!< Peer registering to */
) ;
enum sip_transport transport ; /*!< Transport for this registration UDP, TCP or TLS */
@ -2147,7 +2145,7 @@ static AST_LIST_HEAD_STATIC(threadl, sip_threadinfo);
static struct ao2_container * peers ;
static struct ao2_container * peers_by_ip ;
/*! \brief The register list: Other SIP proxies we register with and place calls to */
/*! \brief The register list: Other SIP proxies we register with and receive calls from */
static struct ast_register_list {
ASTOBJ_CONTAINER_COMPONENTS ( struct sip_registry ) ;
int recheck ;
@ -2322,7 +2320,7 @@ static struct sockaddr_in externip; /*!< External IP address if we are behind N
static char externhost [ MAXHOSTNAMELEN ] ; /*!< External host name */
static time_t externexpire ; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
static int externrefresh = 10 ;
static int externrefresh = 10 ; /*!< Refresh timer for DNS-based external address (dyndns) */
static struct sockaddr_in stunaddr ; /*!< stun server address */
/*! \brief List of local networks
@ -2499,36 +2497,6 @@ static void ast_quiet_chan(struct ast_channel *chan);
static int attempt_transfer ( struct sip_dual * transferer , struct sip_dual * target ) ;
static int do_magic_pickup ( struct ast_channel * channel , const char * extension , const char * context ) ;
/*!
* \ brief 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
*/
# define check_request_transport(peer, tmpl) ({ \
int ret = 0 ; \
if ( peer - > socket . type = = tmpl - > socket . type ) \
; \
else if ( ! ( peer - > transports & tmpl - > socket . type ) ) { \
ast_log ( LOG_ERROR , \
" '%s' is not a valid transport for '%s'. we only use '%s'! ending call. \n " , \
get_transport ( tmpl - > socket . type ) , peer - > name , get_transport_list ( peer - > transports ) \
) ; \
ret = 1 ; \
} else if ( peer - > socket . type & SIP_TRANSPORT_TLS ) { \
ast_log ( LOG_WARNING , \
" peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)! \n " , \
peer - > name , get_transport ( tmpl - > socket . type ) \
) ; \
} else { \
ast_debug ( 1 , \
" peer '%s' has contacted us over %s even though we prefer %s. \n " , \
peer - > name , get_transport ( tmpl - > socket . type ) , get_transport ( peer - > socket . type ) \
) ; \
} \
( ret ) ; \
} )
/*--- Device monitoring and Device/extension state/event handling */
@ -2861,6 +2829,37 @@ static int map_s_x(const struct _map_x_s *table, const char *s, int errorvalue)
}
/*!
* \ brief 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
*/
# define check_request_transport(peer, tmpl) ({ \
int ret = 0 ; \
if ( peer - > socket . type = = tmpl - > socket . type ) \
; \
else if ( ! ( peer - > transports & tmpl - > socket . type ) ) { \
ast_log ( LOG_ERROR , \
" '%s' is not a valid transport for '%s'. we only use '%s'! ending call. \n " , \
get_transport ( tmpl - > socket . type ) , peer - > name , get_transport_list ( peer - > transports ) \
) ; \
ret = 1 ; \
} else if ( peer - > socket . type & SIP_TRANSPORT_TLS ) { \
ast_log ( LOG_WARNING , \
" peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)! \n " , \
peer - > name , get_transport ( tmpl - > socket . type ) \
) ; \
} else { \
ast_debug ( 1 , \
" peer '%s' has contacted us over %s even though we prefer %s. \n " , \
peer - > name , get_transport ( tmpl - > socket . type ) , get_transport ( peer - > socket . type ) \
) ; \
} \
( ret ) ; \
} )
/*! \brief
* duplicate a list of channel variables , \ return the copy .
*/
static struct ast_variable * copy_vars ( struct ast_variable * src )
@ -3875,9 +3874,11 @@ static int __sip_autodestruct(const void *data)
}
}
if ( p - > subscribed = = MWI_NOTIFICATION )
if ( p - > relatedpeer )
if ( p - > subscribed = = MWI_NOTIFICATION ) {
if ( p - > relatedpeer ) {
p - > relatedpeer = unref_peer ( p - > relatedpeer , " __sip_autodestruct: unref peer p->relatedpeer " ) ; /* Remove link to peer. If it's realtime, make sure it's gone from memory) */
}
}
/* Reset schedule ID */
p - > autokillid = - 1 ;
Olle Johansson
16 years ago