[svn-commits] rizzo: branch rizzo/astobj2 r57927 - /team/rizzo/astobj2/channels/chan_sip.c

svn-commits at lists.digium.com svn-commits at lists.digium.com
Mon Mar 5 12:51:43 MST 2007


Author: rizzo
Date: Mon Mar  5 13:51:42 2007
New Revision: 57927

URL: http://svn.digium.com/view/asterisk?view=rev&rev=57927
Log:
merge from trunk up to svn 53098
(also fix a bug in 53092 original line 3162, *inuse = 0
must check for inuse != NULL


Modified:
    team/rizzo/astobj2/channels/chan_sip.c

Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=57927&r1=57926&r2=57927
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Mon Mar  5 13:51:42 2007
@@ -567,7 +567,6 @@
 
 /*! \brief Codecs that we support by default: */
 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
-static int noncodeccapability = AST_RTP_DTMF;
 
 /* Object counters */
 static int suserobjs = 0;                /*!< Static users */
@@ -990,6 +989,7 @@
 	int peercapability;			/*!< Supported peer capability */
 	int prefcodec;				/*!< Preferred codec (outbound only) */
 	int noncodeccapability;			/*!< DTMF RFC2833 telephony-event */
+	int jointnoncodeccapability;            /*!< Joint Non codec capability */
 	int redircodecs;			/*!< Redirect codecs */
 	int maxcallbitrate;			/*!< Maximum Call Bitrate for Video Calls */	
 	struct t38properties t38;		/*!< T38 settings */
@@ -1188,6 +1188,7 @@
 	int capability;			/*!<  Codec capability */
 	int inUse;			/*!< Number of calls in use */
 	int call_limit;			/*!< Limit of concurrent calls */
+	int busy_limit;			/*!< Limit where we signal busy */
 	enum transfermodes allowtransfer;	/*! SIP Refer restriction scheme */
 	struct ast_ha *ha;		/*!<  Access control list */
 	struct ast_variable *chanvars;	/*!<  Variables to set for channel created by user */
@@ -1551,7 +1552,6 @@
 static struct sip_peer *temp_peer(const char *name);
 static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime);
 static struct sip_user *build_user(const char *name, struct ast_variable *v, int realtime);
-static int update_call_counter(struct sip_pvt *fup, int event);
 static void sip_destroy_peer(struct sip_peer *peer);
 static void sip_destroy_user(struct sip_user *user);
 static int sip_poke_peer(struct sip_peer *peer);
@@ -1835,10 +1835,20 @@
 	free(reg);
 }
 
-static void *unref_registry(struct sip_registry *reg)
-{
+static void *registry_unref(struct sip_registry *reg)
+{
+	if (option_debug > 2)
+		ast_log(LOG_DEBUG, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1);
 	ASTOBJ_UNREF(reg, sip_registry_destroy);
 	return NULL;
+}
+
+/*! \brief Add object reference to SIP registry */
+static struct sip_registry *registry_addref(struct sip_registry *reg)
+{
+	if (option_debug > 2)
+		ast_log(LOG_DEBUG, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount + 1);
+	return ASTOBJ_REF(reg); /* Add pointer to registry in packet */
 }
 
 static void registry_destroy_all(void)
@@ -1847,7 +1857,7 @@
 	 * we need to remove them before doing the DESTROYALL
 	 */
 	ASTOBJ_CONTAINER_TRAVERSE(&regl,1, do {
-		unref_registry(iterator);
+		registry_unref(iterator);
 	} while (0));
 
 	ASTOBJ_CONTAINER_DESTROYALL(&regl, sip_registry_destroy);
@@ -2230,7 +2240,7 @@
 		} else {
 			/* If no channel owner, destroy now */
 			/* Let the peerpoke system expire packets when the timer expires for poke_noanswer */
-			if (pkt->method != SIP_OPTIONS)
+			if (pkt->method != SIP_OPTIONS && pkt->method != SIP_REGISTER)
 				set_destroy(pvt);
 		}
 	}
@@ -3028,7 +3038,7 @@
 	do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE );
 
 	if (dialog->rtp) {
-		ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
+		ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
 		ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
 		ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
 		ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
@@ -3283,6 +3293,11 @@
 
 	if (sip_debug_test_pvt(p) || option_debug > 2)
 		ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
+	if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) {
+		update_call_counter(p, DEC_CALL_LIMIT);
+		if (option_debug > 1)
+			ast_log(LOG_DEBUG, "This call did not properly clean up call limits. Call ID %s\n", p->callid);
+	}
 
 	/* Remove link from peer to subscription of MWI */
 	if (p->relatedpeer && p->relatedpeer->mwipvt) /* XXX sip_destroy ? */
@@ -3362,7 +3377,7 @@
 		 */
 		if (p->registry->register_pvt == p)
 			p->registry->register_pvt = pvt_unref(p->registry->register_pvt);
-		p->registry = unref_registry(p->registry);
+		p->registry = registry_unref(p->registry);
 	}
 
 	/* Unlink us from the owner if we have one */
@@ -3428,6 +3443,7 @@
 
 	if (option_debug > 2)
 		ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
+
 	/* Test if we need to check call limits, in order to avoid 
 	   realtime lookups if we do not need it */
 	if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT))
@@ -3455,28 +3471,27 @@
 	switch(event) {
 	/* incoming and outgoing affects the inUse counter */
 	case DEC_CALL_LIMIT:
-		if ( *inuse > 0 ) {
-			if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT))
-				(*inuse)--;
-		} else {
+		/* Decrement inuse count if applicable */
+		if (inuse && ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
+			ast_atomic_fetchadd_int(inuse, -1);
+			ast_clear_flag(&fup->flags[0], SIP_INC_COUNT);
+		} else if (inuse)
 			*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) {
+		/* Decrement ringing count if applicable */
+		if (inringing && ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
+			ast_atomic_fetchadd_int(inringing, -1);
+			ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
+		}
+		/* Decrement onhold count if applicable */
+		if (ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD) && global_notifyhold)
+			sip_peer_hold(fup, FALSE);
+		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 is active and we have reached the limit, reject the call */
 		if (*call_limit > 0 ) {
 			if (*inuse >= *call_limit) {
 				ast_log(LOG_ERROR, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit);
@@ -3489,12 +3504,12 @@
 		}
 		if (inringing && (event == INC_CALL_RINGING)) {
 			if (!ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
-				(*inringing)++;
+				ast_atomic_fetchadd_int(inringing, +1);
 				ast_set_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
 			}
 		}
 		/* Continue */
-		(*inuse)++;
+		ast_atomic_fetchadd_int(inuse, +1);
 		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);
@@ -3502,14 +3517,9 @@
 		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", p->name);
-				ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
-			}
+		if (inringing && ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
+			ast_atomic_fetchadd_int(inringing, -1);
+			ast_clear_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING);
 		}
 		break;
 
@@ -3749,6 +3759,7 @@
 	if (ast->tech_pvt)
 		ast->tech_pvt = pvt_unref(ast->tech_pvt);
 
+	ast_module_unref(ast_module_info->self);
 	/* Do not destroy this pvt until we have timeout or
 	   get an answer to the BYE or INVITE/CANCEL 
 	   If we get no answer during retransmit period, drop the call anyway.
@@ -3941,7 +3952,7 @@
 			   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)
+			if (p->udptl && ast->_state == AST_STATE_UP)
 				res = ast_udptl_write(p->udptl, frame);
 			sip_pvt_unlock(p);
 		}
@@ -4266,6 +4277,7 @@
 	if (!ast_strlen_zero(i->language))
 		ast_string_field_set(tmp, language, i->language);
 	i->owner = tmp;
+	ast_module_ref(ast_module_info->self);
 	ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
 	ast_copy_string(tmp->exten, i->exten, sizeof(tmp->exten));
 
@@ -4660,7 +4672,7 @@
 				ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video" : "", strerror(errno));
 			return __sip_destroy(p);
 		}
-		ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
+		ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
 		ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
 		ast_rtp_settos(p->rtp, global_tos_audio);
 		ast_rtp_set_rtptimeout(p->rtp, global_rtptimeout);
@@ -5006,7 +5018,7 @@
 	reg->callid_valid = FALSE;
 	reg->ocseq = INITIAL_CSEQ;
 	ASTOBJ_CONTAINER_LINK(&regl, reg);	/* Add the new registry entry to the list */
-	unref_registry(reg);		/* release the reference given by ASTOBJ_INIT. The container has another reference */
+	registry_unref(reg);		/* release the reference given by ASTOBJ_INIT. The container has another reference */
 	return 0;
 }
 
@@ -5606,7 +5618,7 @@
 
 	newjointcapability = p->capability & (peercapability | vpeercapability);
 	newpeercapability = (peercapability | vpeercapability);
-	newnoncodeccapability = noncodeccapability & peernoncodeccapability;
+	newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
 		
 		
 	if (debug) {
@@ -5620,7 +5632,7 @@
 			    ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
 
 		ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
-			    ast_rtp_lookup_mime_multiple(s1, BUFSIZ, noncodeccapability, 0, 0),
+			    ast_rtp_lookup_mime_multiple(s1, BUFSIZ, p->noncodeccapability, 0, 0),
 			    ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0, 0),
 			    ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0, 0));
 	}
@@ -5641,7 +5653,7 @@
 		they are acceptable */
 	p->jointcapability = newjointcapability;	/* Our joint codec profile for this call */
 	p->peercapability = newpeercapability;		/* The other sides capability in latest offer */
-	p->noncodeccapability = newnoncodeccapability;	/* DTMF capabilities */
+	p->jointnoncodeccapability = newnoncodeccapability;	/* DTMF capabilities */
 
 	ast_rtp_pt_copy(p->rtp, newaudiortp);
 	if (p->vrtp)
@@ -5683,7 +5695,7 @@
 	if (option_debug > 3)
 		ast_log(LOG_DEBUG, "We have an owner, now see if we need to change this call\n");
 
-	if (!(p->owner->nativeformats & p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
+	if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
 		if (debug) {
 			char s1[BUFSIZ], s2[BUFSIZ];
 			ast_log(LOG_DEBUG, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n", 
@@ -6762,7 +6774,7 @@
 
 	/* Now add DTMF RFC2833 telephony-event as a codec */
 	for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
-		if (!(p->noncodeccapability & x))
+		if (!(p->jointnoncodeccapability & x))
 			continue;
 
 		add_noncodec_to_sdp(p, x, 8000, &m_audio, &a_audio, debug);
@@ -7335,6 +7347,10 @@
 		pidfnote = "Unavailable";
 		break;
 	case AST_EXTENSION_ONHOLD:
+		statestring = "confirmed";
+		local_state = NOTIFY_INUSE;
+		pidfstate = "busy";
+		pidfnote = "On hold";
 		break;
 	case AST_EXTENSION_NOT_INUSE:
 	default:
@@ -7432,6 +7448,11 @@
 		else
 			ast_str_append(&tmp, 0, "<dialog id=\"%s\">\n", p->exten);
 		ast_str_append(&tmp, 0, "<state>%s</state>\n", statestring);
+		if (state == AST_EXTENSION_ONHOLD) {
+			ast_str_append(&tmp, 0, "<local>\n<target uri=\"%s\">\n"
+				"<param pname=\"+sip.rendering\" pvalue=\"no\">\n"
+				"</target>\n</local>\n", mto);
+		}
 		ast_str_append(&tmp, 0, "</dialog>\n</dialog-info>\n");
 		break;
 	case NONE:
@@ -7592,7 +7613,7 @@
 
 		/* decouple the two objects */
 		if (p->registry)
-			p->registry = unref_registry(p->registry);
+			p->registry = registry_unref(p->registry);
 		r->register_pvt = pvt_unref(r->register_pvt);	/* reference goes away */
 	}
 
@@ -7674,7 +7695,7 @@
 			r->portno = ntohs(p->sa.sin_port);
 		ast_set_flag(&p->flags[0], SIP_OUTGOING);	/* Registration is outgoing call */
 		r->register_pvt = pvt_ref(p);			/* Save pointer to SIP dialog */
-		p->registry = ASTOBJ_REF(r);	/* Add pointer to registry in dialog */
+		p->registry = registry_addref(r);	/* Add pointer to registry in dialog */
 		if (!ast_strlen_zero(r->secret))	/* Secret (password) */
 			ast_string_field_set(p, secret, r->secret);
 		if (!ast_strlen_zero(r->md5secret))
@@ -8603,14 +8624,9 @@
 		return;
 
 	/* If they put someone on hold, increment the value... otherwise decrement it */
-	if (hold)
-		peer->onHold++;
-	else
-		peer->onHold--;
-
+	ast_atomic_fetchadd_int(&peer->onHold, (hold ? +1 : -1) );
 	/* Request device state update */
 	ast_device_state_changed("SIP/%s", peer->name);
-
 	return;
 }
 
@@ -10628,6 +10644,8 @@
 		ast_cli(fd, "  VM Extension : %s\n", peer->vmexten);
 		ast_cli(fd, "  LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff);
 		ast_cli(fd, "  Call limit   : %d\n", peer->call_limit);
+		if (peer->busy_limit)
+			ast_cli(fd, "  Busy limit   : %d\n", peer->busy_limit);
 		ast_cli(fd, "  Dynamic      : %s\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)?"Yes":"No"));
 		ast_cli(fd, "  Callerid     : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>"));
 		ast_cli(fd, "  RegisterFrom : %s\n", S_OR(peer->register_from_hdr, ""));
@@ -10717,7 +10735,8 @@
 		astman_append(s, "VoiceMailbox: %s\r\n", peer->mailbox);
 		astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer));
 		astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
-		astman_append(s, "Call limit: %d\r\n", peer->call_limit);
+		astman_append(s, "Call-limit: %d\r\n", peer->call_limit);
+		astman_append(s, "Busy-limit: %d\r\n", peer->busy_limit);
 		astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate);
 		astman_append(s, "Dynamic: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)?"Y":"N"));
 		astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
@@ -10866,6 +10885,7 @@
 {
 	int realtimepeers;
 	int realtimeusers;
+	char codec_buf[BUFSIZ];
 
 	realtimepeers = ast_check_realtime("sippeers");
 	realtimeusers = ast_check_realtime("sipusers");
@@ -10920,6 +10940,9 @@
 	ast_cli(fd, "\nGlobal Signalling Settings:\n");
 	ast_cli(fd, "---------------------------\n");
 	ast_cli(fd, "  Codecs:                 ");
+	ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, global_capability);
+	ast_cli(fd, "%s\n", codec_buf);
+	ast_cli(fd, "  Codec Order:            ");
 	print_codec_to_cli(fd, &default_prefs);
 	ast_cli(fd, "\n");
 	ast_cli(fd, "  T1 minimum:             %d\n", global_t1min);
@@ -12582,7 +12605,7 @@
 	struct sip_registry *r = p->registry;
 
 	r->register_pvt = pvt_unref(r->register_pvt);
-	p->registry = unref_registry(r);
+	p->registry = registry_unref(r);
 	set_destroy(p);
 }
 
@@ -13366,8 +13389,10 @@
 			/* Could not start thread */
 			free(d);	/* We don't need it anymore. If thread is created, d will be free'd
 					   by sip_park_thread() */
+			pthread_attr_destroy(&attr);
 			return 0;
 		}
+		pthread_attr_destroy(&attr);
 	} 
 	return -1;
 }
@@ -14093,7 +14118,7 @@
 		build_contact(p);			/* Build our contact header */
 
 		if (p->rtp) {
-			ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
+			ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
 			ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
 		}
 
@@ -15976,6 +16001,9 @@
 		- registered				AST_DEVICE_NOT_INUSE
 		- fixed IP (!dynamic)			AST_DEVICE_NOT_INUSE
 
+	Peers that does not have a known call and can't be reached by OPTIONS
+		- unreachable                           AST_DEVICE_UNAVAILABLE
+
 	If we return AST_DEVICE_UNKNOWN, the device state engine will try to find
 	out a state by walking the channel list.
 */
@@ -16001,27 +16029,34 @@
 	if ((p = find_peer(host, NULL, 1))) {
 		if (p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) {
 			/* we have an address for the peer */
-			/* if qualify is turned on, check the status */
-			if (p->maxms && (p->lastms > p->maxms)) {
+
+			/* Check status in this order
+				- Hold
+				- Ringing
+				- Busy (enforced only by call limit)
+				- Inuse (we have a call)
+				- Unreachable (qualify)
+			If we don't find any of these state, report AST_DEVICE_NOT_INUSE
+			for registered devices */
+			if (p->onHold) /* First check for hold or ring states */
+				res = AST_DEVICE_ONHOLD;
+			else if (p->inRinging) {
+				if (p->inRinging == p->inUse)
+					res = AST_DEVICE_RINGING;
+				else
+					res = AST_DEVICE_RINGINUSE;
+			} else if (p->call_limit && (p->inUse == p->call_limit)) /* check call limit */
+				res = AST_DEVICE_BUSY;
+			else if (p->call_limit && p->busy_limit && p->inUse >= p->busy_limit)
+				/* We're forcing busy before we've reached the call limit */
+				res = AST_DEVICE_BUSY;
+			else if (p->call_limit && p->inUse)	/* Not busy, but we do have a call */
+				res = AST_DEVICE_INUSE;
+			else if (p->maxms && (p->lastms > p->maxms))
+  	                                 /* We don't have a call. Are we reachable at all? Requires qualify= */
 				res = AST_DEVICE_UNAVAILABLE;
-			} else {
-				/* qualify is not on, or the peer is responding properly */
-				/* check call limit */
-				if (p->call_limit && (p->inUse == p->call_limit))
-					res = AST_DEVICE_BUSY;
-				else if (p->call_limit && p->inUse)
-					res = AST_DEVICE_INUSE;
-				else
-					res = AST_DEVICE_NOT_INUSE;
-				if (p->onHold)
-					res = AST_DEVICE_ONHOLD;
-				else if (p->inRinging) {
-					if (p->inRinging == p->inUse)
-						res = AST_DEVICE_RINGING;
-					else
-						res = AST_DEVICE_RINGINUSE;
-				}
-			}
+			else    /* Default reply if we're registered and have no other data */
+				res = AST_DEVICE_NOT_INUSE;
 		} else {
 			/* there is no address, it's unavailable */
 			res = AST_DEVICE_UNAVAILABLE;
@@ -16664,7 +16699,7 @@
 		if (realtime) {
 			rpeerobjs++;
 			if (option_debug > 2)
-				ast_log(LOG_DEBUG,"-REALTIME- peer built. Name: %s. Peer objects: %d\n", peer->name, rpeerobjs);
+				ast_log(LOG_DEBUG,"-REALTIME- peer built. Name: %s. Peer objects: %d\n", name, rpeerobjs);
 		} else
 			speerobjs++;
 		ASTOBJ_INIT(peer);
@@ -17756,7 +17791,7 @@
 		/* grab a reference to make sure objects do not go away when
 		 * the timeout is canceled.
 		 */
-		ASTOBJ_REF(iterator);
+		registry_addref(iterator);
 		iterator->timeout = ast_sched_add(sched, ms, sip_reregister, iterator);
 		ASTOBJ_UNLOCK(iterator);
 	} while (0)



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