Formatting, doxygen updates
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89555 f38db490-d61c-443f-a65b-d21fe96a405b
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1 changed files with 37 additions and 30 deletions
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@ -239,14 +239,6 @@ enum invitestates {
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INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
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};
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/* Do _NOT_ make any changes to this enum, or the array following it;
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if you think you are doing the right thing, you are probably
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not doing the right thing. If you think there are changes
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needed, get someone else to review them first _before_
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submitting a patch. If these two lists do not match properly
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bad things will happen.
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*/
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enum xmittype {
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XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
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If it fails, it's critical and will cause a teardown of the session */
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@ -284,26 +276,6 @@ static const struct cfsubscription_types {
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{ MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
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};
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/*! \brief SIP Request methods known by Asterisk */
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enum sipmethod {
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SIP_UNKNOWN, /* Unknown response */
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SIP_RESPONSE, /* Not request, response to outbound request */
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SIP_REGISTER,
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SIP_OPTIONS,
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SIP_NOTIFY,
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SIP_INVITE,
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SIP_ACK,
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SIP_PRACK, /* Not supported at all */
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SIP_BYE,
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SIP_REFER,
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SIP_SUBSCRIBE,
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SIP_MESSAGE,
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SIP_UPDATE, /* We can send UPDATE; but not accept it */
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SIP_INFO,
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SIP_CANCEL,
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SIP_PUBLISH, /* Not supported at all */
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SIP_PING, /* Not supported at all, no standard but still implemented out there */
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};
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/*! \brief Authentication types - proxy or www authentication
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\note Endpoints, like Asterisk, should always use WWW authentication to
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@ -324,7 +296,7 @@ enum check_auth_result {
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AUTH_CHALLENGE_SENT = 1,
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AUTH_SECRET_FAILED = -1,
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AUTH_USERNAME_MISMATCH = -2,
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AUTH_NOT_FOUND = -3, /* returned by register_verify */
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AUTH_NOT_FOUND = -3, /*!< returned by register_verify */
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AUTH_FAKE_AUTH = -4,
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AUTH_UNKNOWN_DOMAIN = -5,
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AUTH_PEER_NOT_DYNAMIC = -6,
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@ -379,13 +351,47 @@ struct sip_proxy {
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/* Room for a SRV record chain based on the name */
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};
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/*! \brief States whether a SIP message can create a dialog in Asterisk. */
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enum can_create_dialog {
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CAN_NOT_CREATE_DIALOG,
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CAN_CREATE_DIALOG,
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CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
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};
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/*! XXX Note that sip_methods[i].id == i must hold or the code breaks */
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/*! \brief SIP Request methods known by Asterisk
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\note Do _NOT_ make any changes to this enum, or the array following it;
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if you think you are doing the right thing, you are probably
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not doing the right thing. If you think there are changes
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needed, get someone else to review them first _before_
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submitting a patch. If these two lists do not match properly
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bad things will happen.
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*/
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enum sipmethod {
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SIP_UNKNOWN, /*!< Unknown response */
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SIP_RESPONSE, /*!< Not request, response to outbound request */
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SIP_REGISTER, /*!< Registration to the mothership, tell us where you are located */
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SIP_OPTIONS, /*!< Check capabilities of a device, used for "ping" too */
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SIP_NOTIFY, /*!< Status update, Part of the event package standard, result of a SUBSCRIBE or a REFER */
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SIP_INVITE, /*!< Set up a session */
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SIP_ACK, /*!< End of a three-way handshake started with INVITE. */
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SIP_PRACK, /*!< Reliable pre-call signalling. Not supported in Asterisk. */
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SIP_BYE, /*!< End of a session */
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SIP_REFER, /*!< Refer to another URI (transfer) */
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SIP_SUBSCRIBE, /*!< Subscribe for updates (voicemail, session status, device status, presence) */
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SIP_MESSAGE, /*!< Text messaging */
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SIP_UPDATE, /*!< Update a dialog. We can send UPDATE; but not accept it */
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SIP_INFO, /*!< Information updates during a session */
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SIP_CANCEL, /*!< Cancel an INVITE */
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SIP_PUBLISH, /*!< Not supported in Asterisk */
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SIP_PING, /*!< Not supported at all, no standard but still implemented out there */
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};
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/*! \brief The core structure to setup dialogs. We parse incoming messages by using
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structure and then route the messages according to the type.
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\note Note that sip_methods[i].id == i must hold or the code breaks */
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static const struct cfsip_methods {
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enum sipmethod id;
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int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
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@ -424,6 +430,7 @@ static const struct cfsip_methods {
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#define SUPPORTED 1
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#define NOT_SUPPORTED 0
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/* SIP options */
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#define SIP_OPT_REPLACES (1 << 0)
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#define SIP_OPT_100REL (1 << 1)
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#define SIP_OPT_TIMER (1 << 2)
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