[svn-commits] oej: trunk r151980 - /trunk/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sun Oct 26 04:19:20 CDT 2008
Author: oej
Date: Sun Oct 26 04:19:19 2008
New Revision: 151980
URL: http://svn.digium.com/view/asterisk?view=rev&rev=151980
Log:
Doxygen changes and some formatting.
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=151980&r1=151979&r2=151980
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sun Oct 26 04:19:19 2008
@@ -282,11 +282,16 @@
#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 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 @@
#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 /*!< our initial sip sequence number */
+#define INITIAL_CSEQ 101 /*!< Our 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 @@
.resync_threshold = -1,
.impl = ""
};
-static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration */
-
-static const char config[] = "sip.conf"; /*!< Main configuration file */
+static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration */
+
+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 @@
/*! \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 @@
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 @@
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 @@
#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 */
@@ -4233,12 +4235,8 @@
* 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 @@
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 @@
}
}
- if (!p && (realtime || devstate_only))
+ if (!p && (realtime || devstate_only)) {
p = realtime_peer(peer, sin, devstate_only);
+ }
return p;
}
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