Handle ringing (early) state properly on SIP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Mark Spencer 20 years ago
parent 9d15337fa0
commit fa9e0ed768

@ -510,7 +510,8 @@ static struct io_context *io; /*!< The IO context */
#define DEC_CALL_LIMIT 0
#define INC_CALL_LIMIT 1
#define DEC_CALL_RINGING 2
#define INC_CALL_RINGING 3
/*! \brief sip_request: The data grabbed from the UDP socket */
struct sip_request {
@ -675,7 +676,7 @@ struct sip_auth {
#define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 10) /*!< Allow subscriptions from this peer? */
#define SIP_PAGE2_ALLOWOVERLAP (1 << 11) /*!< Allow overlap dialing ? */
#define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 12) /*!< Only issue MWI notification if subscribed to */
#define SIP_PAGE2_INC_RINGING (1 << 13) /*!< Did this connection increment the counter of in-use calls? */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT)
@ -920,6 +921,7 @@ struct sip_peer {
char cid_name[80]; /*!< Caller ID name */
int callingpres; /*!< Calling id presentation */
int inUse; /*!< Number of calls in use */
int inRinging; /*!< Number of calls ringing */
int call_limit; /*!< Limit of concurrent calls */
enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
@ -2410,7 +2412,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
if (option_debug)
ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username);
res = update_call_counter(p, INC_CALL_LIMIT);
res = update_call_counter(p, INC_CALL_RINGING);
if ( res != -1 ) {
p->callingpres = ast->cid.cid_pres;
p->jointcapability = p->capability;
@ -2560,7 +2562,7 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner)
static int update_call_counter(struct sip_pvt *fup, int event)
{
char name[256];
int *inuse, *call_limit;
int *inuse, *call_limit, *inringing = NULL;
int outgoing = ast_test_flag(&fup->flags[0], SIP_OUTGOING);
struct sip_user *u = NULL;
struct sip_peer *p = NULL;
@ -2589,6 +2591,7 @@ static int update_call_counter(struct sip_pvt *fup, int event)
if (p) {
inuse = &p->inUse;
call_limit = &p->call_limit;
inringing = &p->inRinging;
ast_copy_string(name, fup->peername, sizeof(name));
} else {
if (option_debug > 1)
@ -2605,10 +2608,20 @@ static int update_call_counter(struct sip_pvt *fup, int event)
} else {
*inuse = 0;
}
if (inringing) {
if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
if (*inringing > 0)
(*inringing)--;
else
ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", fup->peername);
ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
}
}
if (option_debug > 1 || sipdebug) {
ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
}
break;
case INC_CALL_RINGING:
case INC_CALL_LIMIT:
if (*call_limit > 0 ) {
if (*inuse >= *call_limit) {
@ -2620,15 +2633,36 @@ static int update_call_counter(struct sip_pvt *fup, int event)
return -1;
}
}
if (inringing && (event == INC_CALL_RINGING)) {
if (!ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
(*inringing)++;
ast_set_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
}
}
/* Continue */
(*inuse)++;
ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
if (option_debug > 1 || sipdebug) {
ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit);
}
break;
case DEC_CALL_RINGING:
if (inringing) {
if (ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
if (*inringing > 0)
(*inringing)--;
else
ast_log(LOG_WARNING, "Inringing for peer '%s' < 0?\n", fup->peername);
ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
}
}
break;
break;
default:
ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
}
if (p)
ast_device_state_changed("SIP/%s", p->name);
if (u)
ASTOBJ_UNREF(u, sip_destroy_user);
else
@ -7827,7 +7861,7 @@ static int sip_show_inuse(int fd, int argc, char *argv[]) {
snprintf(ilimits, sizeof(ilimits), "%d", iterator->call_limit);
else
ast_copy_string(ilimits, "N/A", sizeof(ilimits));
snprintf(iused, sizeof(iused), "%d", iterator->inUse);
snprintf(iused, sizeof(iused), "%d/%d", iterator->inUse, iterator->inRinging);
if (showall || iterator->call_limit)
ast_cli(fd, FORMAT2, iterator->name, iused, ilimits);
ASTOBJ_UNLOCK(iterator);
@ -10042,8 +10076,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
sip_cancel_destroy(p);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
ast_queue_control(p->owner, AST_CONTROL_RINGING);
if (p->owner->_state != AST_STATE_UP)
if (p->owner->_state != AST_STATE_UP) {
ast_setstate(p->owner, AST_STATE_RINGING);
}
}
if (find_sdp(req)) {
process_sdp(p, req);
@ -10076,6 +10111,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
/* When we get 200 OK, we know which device (and IP) to contact for this call */
/* This is important when we have a SIP proxy between us and the phone */
if (outgoing) {
update_call_counter(p, DEC_CALL_RINGING);
parse_ok_contact(p, req);
if(set_address_from_contact(p)) {
/* Bad contact - we don't know how to reach this device */
@ -12503,6 +12539,12 @@ static int sip_devicestate(void *data)
res = AST_DEVICE_INUSE;
else
res = AST_DEVICE_NOT_INUSE;
if (p->inRinging) {
if (p->inRinging == p->inUse)
res = AST_DEVICE_RINGING;
else
res = AST_DEVICE_RINGINUSE;
}
}
} else {
/* there is no address, it's unavailable */

@ -41,6 +41,8 @@ extern "C" {
#define AST_DEVICE_UNAVAILABLE 5
/*! Device is ringing */
#define AST_DEVICE_RINGING 6
/*! Device is ringing *and* in use */
#define AST_DEVICE_RINGINUSE 7
typedef int (*ast_devstate_cb_type)(const char *dev, int state, void *data);

@ -1783,6 +1783,12 @@ static int ast_extension_state2(struct ast_exten *e)
allunavailable = 0;
allfree = 0;
break;
case AST_DEVICE_RINGINUSE:
inuse = 1;
ring = 1;
allunavailable = 0;
allfree = 0;
break;
case AST_DEVICE_BUSY:
allunavailable = 0;
allfree = 0;

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