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- Change "call" to "dialog" where use of "call" is confusing, since it may be a register transaction or a subscription.

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The word dialog is defined as "a peer-to-peer SIP relationship between two UAs that persist for some time" in RFC 3261.

- Whitespace fixes
- Debugging fixes (adding check of option_debug)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2006-01-23 17:08:19 +00:00
parent 814800008d
commit 57f0c81518
1 changed files with 25 additions and 23 deletions
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channels/chan_sip.c
+25 -23
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@@ -395,7 +395,7 @@ AST_MUTEX_DEFINE_STATIC(usecnt_lock);
AST_MUTEX_DEFINE_STATIC(rand_lock);
/*! \brief Protect the interface list (of sip_pvt's) */
/*! \brief Protect the SIP dialog list (of sip_pvt's) */
AST_MUTEX_DEFINE_STATIC(iflock);
/*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
@@ -595,10 +595,10 @@ struct sip_auth {
static int global_rtautoclear = 120;
/*! \brief sip_pvt: PVT structures are used for each SIP conversation, ie. a call */
/*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe */
static struct sip_pvt {
ast_mutex_t lock; /*!< Channel private lock */
int method; /*!< SIP method of this packet */
ast_mutex_t lock; /*!< Dialog private lock */
int method; /*!< SIP method that opened this dialog */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(callid); /*!< Global CallID */
AST_STRING_FIELD(randdata); /*!< Random data */
@@ -690,35 +690,35 @@ static struct sip_pvt {
int rtptimeout; /*!< RTP timeout time */
int rtpholdtimeout; /*!< RTP timeout when on hold */
int rtpkeepalive; /*!< Send RTP packets for keepalive */
enum subscriptiontype subscribed; /*!< Is this call a subscription? */
enum subscriptiontype subscribed; /*!< Is this dialog a subscription? */
int stateid;
int laststate; /*!< Last known extension state */
int dialogver;
struct ast_dsp *vad; /*!< Voice Activation Detection dsp */
struct sip_peer *peerpoke; /*!< If this calls is to poke a peer, which one */
struct sip_registry *registry; /*!< If this is a REGISTER call, to which registry */
struct sip_peer *peerpoke; /*!< If this dialog is to poke a peer, which one */
struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */
struct ast_rtp *rtp; /*!< RTP Session */
struct ast_rtp *vrtp; /*!< Video RTP session */
struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
struct sip_history_head *history; /*!< History of this SIP dialog */
struct ast_variable *chanvars; /*!< Channel variables to set for call */
struct sip_pvt *next; /*!< Next call in chain */
struct sip_pvt *next; /*!< Next dialog in chain */
struct sip_invite_param *options; /*!< Options for INVITE */
} *iflist = NULL;
#define FLAG_RESPONSE (1 << 0)
#define FLAG_FATAL (1 << 1)
/*! \brief sip packet - read in sipsock_read, transmitted in send_request */
/*! \brief sip packet - read in sipsock_read(), transmitted in send_request() */
struct sip_pkt {
struct sip_pkt *next; /*!< Next packet */
int retrans; /*!< Retransmission number */
int method; /*!< SIP method for this packet */
int seqno; /*!< Sequence number */
unsigned int flags; /*!< non-zero if this is a response packet (e.g. 200 OK) */
struct sip_pvt *owner; /*!< Owner call */
struct sip_pvt *owner; /*!< Owner AST call */
int retransid; /*!< Retransmission ID */
int timer_a; /*!< SIP timer A, retransmission timer */
int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */
@@ -848,7 +848,7 @@ struct sip_registry {
int regattempts; /*!< Number of attempts (since the last success) */
int timeout; /*!< sched id of sip_reg_timeout */
int refresh; /*!< How often to refresh */
struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration call" in progress */
struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
int regstate; /*!< Registration state (see above) */
int callid_valid; /*!< 0 means we haven't chosen callid for this registry yet. */
unsigned int ocseq; /*!< Sequence number we got to for REGISTERs for this registry */
@@ -867,7 +867,7 @@ static struct ast_peer_list {
ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer);
} peerl;
/*! \brief The register list: Other SIP proxys we register with and call */
/*! \brief The register list: Other SIP proxys we register with and place calls to */
static struct ast_register_list {
ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
int recheck;
@@ -1310,7 +1310,7 @@ static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *dat
return 0;
}
/*! \brief __sip_autodestruct: Kill a call (called by scheduler) */
/*! \brief __sip_autodestruct: Kill a SIP dialog (called by scheduler) */
static int __sip_autodestruct(void *data)
{
struct sip_pvt *p = data;
@@ -1328,7 +1328,7 @@ static int __sip_autodestruct(void *data)
ast_log(LOG_DEBUG, "Auto destroying call '%s'\n", p->callid);
append_history(p, "AutoDestroy", "");
if (p->owner) {
ast_log(LOG_WARNING, "Autodestruct on call '%s' with owner in place\n", p->callid);
ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
ast_queue_hangup(p->owner);
} else {
sip_destroy(p);
@@ -1340,7 +1340,7 @@ static int __sip_autodestruct(void *data)
static int sip_scheddestroy(struct sip_pvt *p, int ms)
{
if (sip_debug_test_pvt(p))
ast_verbose("Scheduling destruction of call '%s' in %d ms\n", p->callid, ms);
ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
if (recordhistory)
append_history(p, "SchedDestroy", "%d ms", ms);
@@ -1350,7 +1350,7 @@ static int sip_scheddestroy(struct sip_pvt *p, int ms)
return 0;
}
/*! \brief sip_cancel_destroy: Cancel destruction of SIP call */
/*! \brief sip_cancel_destroy: Cancel destruction of SIP dialog */
static int sip_cancel_destroy(struct sip_pvt *p)
{
if (p->autokillid > -1)
@@ -2042,7 +2042,8 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
#ifdef OSP_SUPPORT
if (!p->options->osptoken || !osphandle || (sscanf(osphandle, "%d", &p->osphandle) != 1)) {
/* Force Disable OSP support */
ast_log(LOG_DEBUG, "Disabling OSP support for this call. osptoken = %s, osphandle = %s\n", p->options->osptoken, osphandle);
if (option_debug)
ast_log(LOG_DEBUG, "Disabling OSP support for this call. osptoken = %s, osphandle = %s\n", p->options->osptoken, osphandle);
p->options->osptoken = NULL;
osphandle = NULL;
p->osphandle = -1;
@@ -2083,14 +2084,14 @@ static void sip_registry_destroy(struct sip_registry *reg)
}
/*! \brief __sip_destroy: Execute destrucion of call structure, release memory*/
/*! \brief __sip_destroy: Execute destrucion of SIP dialog structure, release memory */
static void __sip_destroy(struct sip_pvt *p, int lockowner)
{
struct sip_pvt *cur, *prev = NULL;
struct sip_pkt *cp;
if (sip_debug_test_pvt(p))
ast_verbose("Destroying call '%s'\n", p->callid);
ast_verbose("Destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
if (dumphistory)
sip_dump_history(p);
@@ -2118,14 +2119,15 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner)
if (p->registry) {
if (p->registry->call == p)
p->registry->call = NULL;
ASTOBJ_UNREF(p->registry,sip_registry_destroy);
ASTOBJ_UNREF(p->registry, sip_registry_destroy);
}
/* Unlink us from the owner if we have one */
if (p->owner) {
if (lockowner)
ast_mutex_lock(&p->owner->lock);
ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
if (option_debug)
ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
p->owner->tech_pvt = NULL;
if (lockowner)
ast_mutex_unlock(&p->owner->lock);
@@ -2440,7 +2442,7 @@ static char *hangup_cause2sip(int cause)
}
/*! \brief sip_hangup: Hangup SIP call
/*! \brief sip_hangup: Hangup SIP call
* Part of PBX interface, called from ast_hangup */
static int sip_hangup(struct ast_channel *ast)
{
@@ -2733,7 +2735,7 @@ static int sip_indicate(struct ast_channel *ast, int condition)
break;
case AST_CONTROL_HOLD: /* The other part of the bridge are put on hold */
if (sipdebug)
ast_log(LOG_DEBUG, "Bridged channel now on hold%s\n", p->callid);
ast_log(LOG_DEBUG, "Bridged channel now on hold - %s\n", p->callid);
res = -1;
break;
case AST_CONTROL_UNHOLD: /* The other part of the bridge are back from hold */