- Doxygen additions
- Remove unused string in sip_registry -- "random" - Someone added a function in the middle of all forward declarations... Weird. Moved it out of that section. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@216905 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -1152,13 +1152,13 @@ static const char *sip_reason_code_to_str(enum AST_REDIRECTING_REASON code)
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configuring devices
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*/
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/*@{*/
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static char default_language[MAX_LANGUAGE]; /*! Default language setting for new channels */
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static char default_callerid[AST_MAX_EXTENSION];
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static char default_mwi_from[80];
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static char default_fromdomain[AST_MAX_EXTENSION];
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static char default_notifymime[AST_MAX_EXTENSION];
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static char default_language[MAX_LANGUAGE]; /*!< Default language setting for new channels */
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static char default_callerid[AST_MAX_EXTENSION]; /*!< Default caller ID for sip messages */
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static char default_mwi_from[80]; /*!< Default caller ID for MWI updates */
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static char default_fromdomain[AST_MAX_EXTENSION]; /*!< Default domain on outound messages */
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static char default_notifymime[AST_MAX_EXTENSION]; /*!< Default MIME media type for MWI notify messages */
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static char default_vmexten[AST_MAX_EXTENSION]; /*!< Default From Username on MWI updates */
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static int default_qualify; /*!< Default Qualify= setting */
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static char default_vmexten[AST_MAX_EXTENSION];
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static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
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static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
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* a bridged channel on hold */
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@ -1177,6 +1177,7 @@ static unsigned int default_primary_transport; /*!< Default primary Transport (
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*/
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/*@{*/
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/*! \brief a place to store all global settings for the sip channel driver
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These are settings that will be possibly to apply on a group level later on.
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\note Do not add settings that only apply to the channel itself and can't
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be applied to devices (trunks, services, phones)
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@ -1342,10 +1343,8 @@ struct sip_request {
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char debug; /*!< print extra debugging if non zero */
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char has_to_tag; /*!< non-zero if packet has To: tag */
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char ignore; /*!< if non-zero This is a re-transmit, ignore it */
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/* Array of offsets into the request string of each SIP header*/
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ptrdiff_t header[SIP_MAX_HEADERS];
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/* Array of offsets into the request string of each SDP line*/
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ptrdiff_t line[SIP_MAX_LINES];
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ptrdiff_t header[SIP_MAX_HEADERS]; /*!< Array of offsets into the request string of each SIP header*/
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ptrdiff_t line[SIP_MAX_LINES]; /*!< Array of offsets into the request string of each SDP line*/
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struct ast_str *data;
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/* XXX Do we need to unref socket.ser when the request goes away? */
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struct sip_socket socket; /*!< The socket used for this request */
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@ -1845,10 +1844,11 @@ struct sip_pvt {
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* By doing this, even if we don't want to answer a particular media stream with something meaningful, we can
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* still put an m= line in our answer with the port set to 0.
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*
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* The reason for the length being 4 is that in this branch of Asterisk, the only media types supported are
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* The reason for the length being 4 (OFFERED_MEDIA_COUNT) is that in this branch of Asterisk, the only media types supported are
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* image, audio, text, and video. Therefore we need to keep track of which types of media were offered.
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* Note that secure RTP defines new types of SDP media.
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*
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* Note that if we wanted to be 100% correct, we would keep a list of all media streams offered. That way we could respond
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* If we wanted to be 100% correct, we would keep a list of all media streams offered. That way we could respond
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* even to unknown media types, and we could respond to multiple streams of the same type. Such large-scale changes
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* are not a good idea for released branches, though, so we're compromising by just making sure that for the common cases:
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* audio and video, audio and T.38, and audio and text, we give the appropriate response to both media streams.
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@ -2016,7 +2016,7 @@ struct sip_peer {
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/*! Mailboxes that this peer cares about */
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AST_LIST_HEAD_NOLOCK(, sip_mailbox) mailboxes;
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int maxcallbitrate; /*!< Maximum Bitrate for a video call */
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int maxcallbitrate; /*!< Maximum Bitrate for a video call */
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int expire; /*!< When to expire this peer registration */
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int capability; /*!< Codec capability */
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int rtptimeout; /*!< RTP timeout */
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@ -2027,13 +2027,12 @@ struct sip_peer {
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struct sip_proxy *outboundproxy; /*!< Outbound proxy for this peer */
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struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
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struct sockaddr_in addr; /*!< IP address of peer */
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/* Qualification */
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struct sip_pvt *call; /*!< Call pointer */
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int pokeexpire; /*!< When to expire poke (qualify= checking) */
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int lastms; /*!< How long last response took (in ms), or -1 for no response */
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int maxms; /*!< Max ms we will accept for the host to be up, 0 to not monitor */
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int qualifyfreq; /*!< Qualification: How often to check for the host to be up */
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struct timeval ps; /*!< Time for sending SIP OPTION in sip_pke_peer() */
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int pokeexpire; /*!< Qualification: When to expire poke (qualify= checking) */
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int lastms; /*!< Qualification: How long last response took (in ms), or -1 for no response */
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int maxms; /*!< Qualification: Max ms we will accept for the host to be up, 0 to not monitor */
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int qualifyfreq; /*!< Qualification: Qualification: How often to check for the host to be up */
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struct timeval ps; /*!< Qualification: Time for sending SIP OPTION in sip_pke_peer() */
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struct sockaddr_in defaddr; /*!< Default IP address, used until registration */
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struct ast_ha *ha; /*!< Access control list */
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struct ast_ha *contactha; /*!< Restrict what IPs are allowed in the Contact header (for registration) */
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@ -2078,7 +2077,6 @@ struct sip_registry {
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AST_STRING_FIELD(secret); /*!< Password in clear text */
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AST_STRING_FIELD(md5secret); /*!< Password in md5 */
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AST_STRING_FIELD(callback); /*!< Contact extension */
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AST_STRING_FIELD(random);
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AST_STRING_FIELD(peername); /*!< Peer registering to */
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);
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enum sip_transport transport; /*!< Transport for this registration UDP, TCP or TLS */
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@ -2147,7 +2145,7 @@ static AST_LIST_HEAD_STATIC(threadl, sip_threadinfo);
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static struct ao2_container *peers;
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static struct ao2_container *peers_by_ip;
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/*! \brief The register list: Other SIP proxies we register with and place calls to */
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/*! \brief The register list: Other SIP proxies we register with and receive calls from */
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static struct ast_register_list {
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ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
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int recheck;
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@ -2322,7 +2320,7 @@ static struct sockaddr_in externip; /*!< External IP address if we are behind N
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static char externhost[MAXHOSTNAMELEN]; /*!< External host name */
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static time_t externexpire; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
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static int externrefresh = 10;
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static int externrefresh = 10; /*!< Refresh timer for DNS-based external address (dyndns) */
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static struct sockaddr_in stunaddr; /*!< stun server address */
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/*! \brief List of local networks
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@ -2499,36 +2497,6 @@ static void ast_quiet_chan(struct ast_channel *chan);
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static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
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static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
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/*!
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* \brief generic function for determining if a correct transport is being
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* used to contact a peer
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*
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* this is done as a macro so that the "tmpl" var can be passed either a
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* sip_request or a sip_peer
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*/
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#define check_request_transport(peer, tmpl) ({ \
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int ret = 0; \
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if (peer->socket.type == tmpl->socket.type) \
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; \
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else if (!(peer->transports & tmpl->socket.type)) {\
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ast_log(LOG_ERROR, \
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"'%s' is not a valid transport for '%s'. we only use '%s'! ending call.\n", \
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get_transport(tmpl->socket.type), peer->name, get_transport_list(peer->transports) \
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); \
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ret = 1; \
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} else if (peer->socket.type & SIP_TRANSPORT_TLS) { \
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ast_log(LOG_WARNING, \
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"peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)!\n", \
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peer->name, get_transport(tmpl->socket.type) \
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); \
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} else { \
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ast_debug(1, \
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"peer '%s' has contacted us over %s even though we prefer %s.\n", \
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peer->name, get_transport(tmpl->socket.type), get_transport(peer->socket.type) \
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); \
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}\
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(ret); \
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})
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/*--- Device monitoring and Device/extension state/event handling */
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@ -2861,6 +2829,37 @@ static int map_s_x(const struct _map_x_s *table, const char *s, int errorvalue)
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}
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/*!
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* \brief generic function for determining if a correct transport is being
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* used to contact a peer
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*
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* this is done as a macro so that the "tmpl" var can be passed either a
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* sip_request or a sip_peer
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*/
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#define check_request_transport(peer, tmpl) ({ \
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int ret = 0; \
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if (peer->socket.type == tmpl->socket.type) \
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; \
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else if (!(peer->transports & tmpl->socket.type)) {\
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ast_log(LOG_ERROR, \
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"'%s' is not a valid transport for '%s'. we only use '%s'! ending call.\n", \
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get_transport(tmpl->socket.type), peer->name, get_transport_list(peer->transports) \
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); \
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ret = 1; \
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} else if (peer->socket.type & SIP_TRANSPORT_TLS) { \
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ast_log(LOG_WARNING, \
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"peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)!\n", \
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peer->name, get_transport(tmpl->socket.type) \
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); \
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} else { \
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ast_debug(1, \
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"peer '%s' has contacted us over %s even though we prefer %s.\n", \
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peer->name, get_transport(tmpl->socket.type), get_transport(peer->socket.type) \
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); \
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}\
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(ret); \
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})
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/*! \brief
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* duplicate a list of channel variables, \return the copy.
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*/
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static struct ast_variable *copy_vars(struct ast_variable *src)
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@ -3875,9 +3874,11 @@ static int __sip_autodestruct(const void *data)
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}
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}
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if (p->subscribed == MWI_NOTIFICATION)
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if (p->relatedpeer)
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if (p->subscribed == MWI_NOTIFICATION) {
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if (p->relatedpeer) {
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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) */
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}
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}
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/* Reset schedule ID */
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p->autokillid = -1;
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