[asterisk-commits] oej: branch 1.4 r48177 - in /branches/1.4: channels/ configs/

[asterisk-commits at lists.digium.com]

Author: oej
Date: Fri Dec  1 11:41:56 2006
New Revision: 48177

URL: http://svn.digium.com/view/asterisk?view=rev&rev=48177
Log:
- Backport of the "limitonpeers" patch from trunk, to fix a lot of issues with queues and SIP device states
- Remove support for T.38 early media, since it's impossible.

(Two patches in one - extra friday evening offer due to being off line from svn today... :-)

Modified:
    branches/1.4/channels/chan_sip.c
    branches/1.4/configs/sip.conf.sample

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=48177&r1=48176&r2=48177
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Fri Dec  1 11:41:56 2006
@@ -513,6 +513,7 @@
 static struct ast_codec_pref default_prefs;		/*!< Default codec prefs */
 
 /* Global settings only apply to the channel */
+static int global_limitonpeers;		/*!< Match call limit on peers only */
 static int global_rtautoclear;
 static int global_notifyringing;	/*!< Send notifications on ringing */
 static int global_notifyhold;		/*!< Send notifications on hold */
@@ -2944,7 +2945,7 @@
 static int update_call_counter(struct sip_pvt *fup, int event)
 {
 	char name[256];
-	int *inuse, *call_limit, *inringing;
+	int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
 	int outgoing = ast_test_flag(&fup->flags[0], SIP_OUTGOING);
 	struct sip_user *u = NULL;
 	struct sip_peer *p = NULL;
@@ -2959,16 +2960,17 @@
 	ast_copy_string(name, fup->username, sizeof(name));
 
 	/* Check the list of users only for incoming calls */
-	if (!outgoing && (u = find_user(name, 1)) ) {
+	if (global_limitonpeers == FALSE && !outgoing && (u = find_user(name, 1)))  {
 		inuse = &u->inUse;
 		call_limit = &u->call_limit;
 		inringing = NULL;
-	} else if ( (p = find_peer(fup->peername, NULL, 1) ) ) { /* Try to find peer */
+	} else if ( (p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, 1) ) ) { /* Try to find peer */
 		inuse = &p->inUse;
 		call_limit = &p->call_limit;
 		inringing = &p->inRinging;
 		ast_copy_string(name, fup->peername, sizeof(name));
-	} else {
+	} 
+	if (!p && !u) {
 		if (option_debug > 1)
 			ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name);
 		return 0;
@@ -3469,15 +3471,12 @@
 	case AST_FRAME_MODEM:
 		if (p) {
 			ast_mutex_lock(&p->lock);
-			if (p->udptl) {
-				if ((ast->_state != AST_STATE_UP) &&
-					!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) && 
-				    !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
-					transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
-					ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
-				}
+			/* UDPTL requires two-way communication, so early media is not needed here.
+				we simply forget the frames if we get modem frames before the bridge is up.
+				Fax will re-transmit.
+			*/
+			if (p->udptl && ast->_state != AST_STATE_UP) 
 				res = ast_udptl_write(p->udptl, frame);
-			}
 			ast_mutex_unlock(&p->lock);
 		}
 		break;
@@ -10155,6 +10154,7 @@
 	ast_cli(fd, "  Our auth realm          %s\n", global_realm);
 	ast_cli(fd, "  Realm. auth:            %s\n", authl ? "Yes": "No");
  	ast_cli(fd, "  Always auth rejects:    %s\n", global_alwaysauthreject ? "Yes" : "No");
+	ast_cli(fd, "  Call limit peers only:  %s\n", global_limitonpeers ? "Yes" : "No");
 	ast_cli(fd, "  User Agent:             %s\n", global_useragent);
 	ast_cli(fd, "  MWI checking interval:  %d secs\n", global_mwitime);
 	ast_cli(fd, "  Reg. context:           %s\n", S_OR(global_regcontext, "(not set)"));
@@ -15892,6 +15892,7 @@
 	global_regcontext[0] = '\0';
 	expiry = DEFAULT_EXPIRY;
 	global_notifyringing = DEFAULT_NOTIFYRINGING;
+	global_limitonpeers = FALSE;
 	global_notifyhold = FALSE;
 	global_alwaysauthreject = 0;
 	global_allowsubscribe = FALSE;
@@ -16014,6 +16015,8 @@
 			compactheaders = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "notifymimetype")) {
 			ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime));
+		} else if (!strcasecmp(v->name, "limitonpeers")) {
+			global_limitonpeers = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "notifyringing")) {
 			global_notifyringing = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "notifyhold")) {

Modified: branches/1.4/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.4/configs/sip.conf.sample?view=diff&rev=48177&r1=48176&r2=48177
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Fri Dec  1 11:41:56 2006
@@ -185,6 +185,12 @@
 ;notifyhold = yes		; Notify subscriptions on HOLD state (default: no)
 				; Turning on notifyringing and notifyhold will add a lot
 				; more database transactions if you are using realtime.
+;limitonpeer = yes		; Apply call limits on peers only. This will improve 
+				; status notification when you are using type=friend
+				; Inbound calls, that really apply to the user part
+				; of a friend will now be added to and compared with
+				; the peer limit instead of applying two call limits,
+				; one for the peer and one for the user.
 
 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
 ;