[svn-commits] trunk r29619 - in /trunk: ./ channels/
include/asterisk/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Mon May 22 21:49:31 MST 2006
Author: markster
Date: Mon May 22 23:49:31 2006
New Revision: 29619
URL: http://svn.digium.com/view/asterisk?rev=29619&view=rev
Log:
Handle ringing (early) state properly on SIP
Modified:
trunk/channels/chan_sip.c
trunk/include/asterisk/devicestate.h
trunk/pbx.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=29619&r1=29618&r2=29619&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon May 22 23:49:31 2006
@@ -510,7 +510,8 @@
#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 @@
#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 @@
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 @@
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 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 @@
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 @@
} 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 @@
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 @@
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 @@
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 @@
/* 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 @@
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 */
Modified: trunk/include/asterisk/devicestate.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/devicestate.h?rev=29619&r1=29618&r2=29619&view=diff
==============================================================================
--- trunk/include/asterisk/devicestate.h (original)
+++ trunk/include/asterisk/devicestate.h Mon May 22 23:49:31 2006
@@ -41,6 +41,8 @@
#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);
Modified: trunk/pbx.c
URL: http://svn.digium.com/view/asterisk/trunk/pbx.c?rev=29619&r1=29618&r2=29619&view=diff
==============================================================================
--- trunk/pbx.c (original)
+++ trunk/pbx.c Mon May 22 23:49:31 2006
@@ -1779,6 +1779,12 @@
allfree = 0;
break;
case AST_DEVICE_RINGING:
+ ring = 1;
+ allunavailable = 0;
+ allfree = 0;
+ break;
+ case AST_DEVICE_RINGINUSE:
+ inuse = 1;
ring = 1;
allunavailable = 0;
allfree = 0;
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