[asterisk-commits] trunk r36180 - /trunk/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Jun 27 12:28:05 MST 2006
Author: oej
Date: Tue Jun 27 14:28:04 2006
New Revision: 36180
URL: http://svn.digium.com/view/asterisk?rev=36180&view=rev
Log:
Formatting and doxygen fixes
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=36180&r1=36179&r2=36180&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Jun 27 14:28:04 2006
@@ -828,9 +828,9 @@
char referred_by[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
char referred_by_name[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
char refer_contact[AST_MAX_EXTENSION]; /*!< Place to store Contact info from a REFER extension */
- char replaces_callid[BUFSIZ]; /*!< Replace info */
- char replaces_callid_totag[BUFSIZ/2]; /*!< Replace info */
- char replaces_callid_fromtag[BUFSIZ/2]; /*!< Replace info */
+ char replaces_callid[BUFSIZ]; /*!< Replace info: callid */
+ char replaces_callid_totag[BUFSIZ/2]; /*!< Replace info: to-tag */
+ char replaces_callid_fromtag[BUFSIZ/2]; /*!< Replace info: from-tag */
struct sip_pvt *refer_call; /*!< Call we are referring */
int attendedtransfer; /*!< Attended or blind transfer? */
int localtransfer; /*!< Transfer to local domain? */
@@ -871,15 +871,14 @@
AST_STRING_FIELD(okcontacturi); /*!< URI from the 200 OK on INVITE */
AST_STRING_FIELD(peersecret); /*!< Password */
AST_STRING_FIELD(peermd5secret);
- AST_STRING_FIELD(cid_num); /*!< Caller*ID */
- AST_STRING_FIELD(cid_name); /*!< Caller*ID */
+ AST_STRING_FIELD(cid_num); /*!< Caller*ID number */
+ AST_STRING_FIELD(cid_name); /*!< Caller*ID name */
AST_STRING_FIELD(via); /*!< Via: header */
AST_STRING_FIELD(fullcontact); /*!< The Contact: that the UA registers with us */
AST_STRING_FIELD(our_contact); /*!< Our contact header */
AST_STRING_FIELD(rpid); /*!< Our RPID header */
AST_STRING_FIELD(rpid_from); /*!< Our RPID From header */
);
- struct ast_codec_pref prefs; /*!< codec prefs */
unsigned int ocseq; /*!< Current outgoing seqno */
unsigned int icseq; /*!< Current incoming seqno */
ast_group_t callgroup; /*!< Call group */
@@ -887,7 +886,8 @@
int lastinvite; /*!< Last Cseq of invite */
struct ast_flags flags[2]; /*!< SIP_ flags */
int timer_t1; /*!< SIP timer T1, ms rtt */
- unsigned int sipoptions; /*!< Supported SIP sipoptions on the other end */
+ unsigned int sipoptions; /*!< Supported SIP options on the other end */
+ struct ast_codec_pref prefs; /*!< codec prefs */
int capability; /*!< Special capability (codec) */
int jointcapability; /*!< Supported capability at both ends (codecs ) */
int peercapability; /*!< Supported peer capability */
@@ -901,40 +901,40 @@
int callingpres; /*!< Calling presentation */
int authtries; /*!< Times we've tried to authenticate */
int expiry; /*!< How long we take to expire */
- long branch; /*!< One random number */
- char tag[11]; /*!< Another random number */
+ long branch; /*!< The branch identifier of this session */
+ char tag[11]; /*!< Our tag for this session */
int sessionid; /*!< SDP Session ID */
int sessionversion; /*!< SDP Session Version */
struct sockaddr_in sa; /*!< Our peer */
struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
+ time_t lastrtprx; /*!< Last RTP received */
+ time_t lastrtptx; /*!< Last RTP sent */
+ int rtptimeout; /*!< RTP timeout time */
+ int rtpholdtimeout; /*!< RTP timeout when on hold */
+ int rtpkeepalive; /*!< Send RTP packets for keepalive */
struct sockaddr_in recv; /*!< Received as */
struct in_addr ourip; /*!< Our IP */
- struct ast_channel *owner; /*!< Who owns us */
+ struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
struct sip_route *route; /*!< Head of linked list of routing steps (fm Record-Route) */
int route_persistant; /*!< Is this the "real" route? */
struct sip_auth *peerauth; /*!< Realm authentication */
int noncecount; /*!< Nonce-count */
char lastmsg[256]; /*!< Last Message sent/received */
int amaflags; /*!< AMA Flags */
- int pendinginvite; /*!< Any pending invite */
+ int pendinginvite; /*!< Any pending invite ? (seqno of this) */
struct sip_request initreq; /*!< Initial request that opened the SIP dialog */
int maxtime; /*!< Max time for first response */
- int initid; /*!< Auto-congest ID if appropriate */
- int autokillid; /*!< Auto-kill ID */
- time_t lastrtprx; /*!< Last RTP received */
- time_t lastrtptx; /*!< Last RTP sent */
- int rtptimeout; /*!< RTP timeout time */
- int rtpholdtimeout; /*!< RTP timeout when on hold */
- int rtpkeepalive; /*!< Send RTP packets for keepalive */
- enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
+ int initid; /*!< Auto-congest ID if appropriate (scheduler) */
+ int autokillid; /*!< Auto-kill ID (scheduler) */
+ enum transfermodes allowtransfer; /*!< REFER: restriction scheme */
+ struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
int stateid; /*!< SUBSCRIBE: ID for devicestate subscriptions */
int laststate; /*!< SUBSCRIBE: Last known extension state */
int dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */
- struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
struct ast_dsp *vad; /*!< Voice Activation Detection dsp */
struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one
@@ -14372,6 +14372,7 @@
fastrestart = FALSE;
curpeernum = 0;
peer = NULL;
+ /* Find next peer that needs mwi */
ASTOBJ_CONTAINER_TRAVERSE(&peerl, !peer, do {
if ((curpeernum > lastpeernum) && does_peer_need_mwi(iterator)) {
fastrestart = TRUE;
@@ -14381,6 +14382,7 @@
curpeernum++;
} while (0)
);
+ /* Send MWI to the peer */
if (peer) {
ASTOBJ_WRLOCK(peer);
sip_send_mwi_to_peer(peer);
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