[asterisk-commits] oej: branch oej/pinequeue-1.8 r369920 - in /team/oej/pinequeue-1.8: ./ apps/ ...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Jul 11 07:05:17 CDT 2012


Author: oej
Date: Wed Jul 11 07:05:12 2012
New Revision: 369920

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369920
Log:
Resolving conflicts

Added:
    team/oej/pinequeue-1.8/main/adsi.c
      - copied unchanged from r369869, branches/1.8/main/adsi.c
Removed:
    team/oej/pinequeue-1.8/res/res_adsi.exports.in
Modified:
    team/oej/pinequeue-1.8/   (props changed)
    team/oej/pinequeue-1.8/apps/app_dial.c
    team/oej/pinequeue-1.8/apps/app_queue.c
    team/oej/pinequeue-1.8/apps/app_stack.c
    team/oej/pinequeue-1.8/apps/app_voicemail.c
    team/oej/pinequeue-1.8/channels/chan_sip.c
    team/oej/pinequeue-1.8/channels/sip/include/sip.h
    team/oej/pinequeue-1.8/configs/sip_notify.conf.sample
    team/oej/pinequeue-1.8/include/asterisk/adsi.h
    team/oej/pinequeue-1.8/main/Makefile
    team/oej/pinequeue-1.8/main/bridging.c
    team/oej/pinequeue-1.8/main/cdr.c
    team/oej/pinequeue-1.8/main/features.c
    team/oej/pinequeue-1.8/main/pbx.c
    team/oej/pinequeue-1.8/main/tcptls.c
    team/oej/pinequeue-1.8/res/res_adsi.c

Propchange: team/oej/pinequeue-1.8/
------------------------------------------------------------------------------
    automerge = Is-there-life-off-net?

Propchange: team/oej/pinequeue-1.8/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed Jul 11 07:05:12 2012
@@ -1,1 +1,1 @@
-/branches/1.8:1-369250
+/branches/1.8:1-369914

Modified: team/oej/pinequeue-1.8/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequeue-1.8/apps/app_dial.c?view=diff&rev=369920&r1=369919&r2=369920
==============================================================================
--- team/oej/pinequeue-1.8/apps/app_dial.c (original)
+++ team/oej/pinequeue-1.8/apps/app_dial.c Wed Jul 11 07:05:12 2012
@@ -2688,7 +2688,9 @@
 			ast_copy_string(peer->context, chan->context, sizeof(peer->context));
 			ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
 			peer->priority = chan->priority + 2;
-			ast_pbx_start(peer);
+			if (ast_pbx_start(peer)) {
+				ast_hangup(peer);
+			}
 			hanguptree(outgoing, NULL, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
 			if (continue_exec)
 				*continue_exec = 1;
@@ -2979,24 +2981,26 @@
 			}
 			ast_set2_flag(peer, autoloopflag, AST_FLAG_IN_AUTOLOOP);  /* set it back the way it was */
 		}
-		if (!ast_check_hangup(peer) && ast_test_flag64(&opts, OPT_CALLEE_GO_ON)) {
-			if(!ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GO_ON])) {
-				replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GO_ON]);
-				ast_parseable_goto(peer, opt_args[OPT_ARG_CALLEE_GO_ON]);
-			} else { /* F() */
-				int res;
-				res = ast_goto_if_exists(peer, chan->context, chan->exten, (chan->priority) + 1); 
-				if (res == AST_PBX_GOTO_FAILED) {
-					ast_hangup(peer);
+		if (!ast_check_hangup(peer)) {
+			if (ast_test_flag64(&opts, OPT_CALLEE_GO_ON)) {
+				int goto_res;
+
+				if (!ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GO_ON])) {
+					replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GO_ON]);
+					goto_res = ast_parseable_goto(peer, opt_args[OPT_ARG_CALLEE_GO_ON]);
+				} else { /* F() */
+					goto_res = ast_goto_if_exists(peer, chan->context,
+						chan->exten, chan->priority + 1);
+				}
+				if (!goto_res && !ast_pbx_start(peer)) {
+					/* The peer is now running its own PBX. */
 					goto out;
 				}
-			}
-			ast_pbx_start(peer);
-		} else {
-			if (!ast_check_hangup(chan))
+			} else {
 				chan->hangupcause = peer->hangupcause;
-			ast_hangup(peer);
-		}
+			}
+		}
+		ast_hangup(peer);
 	}
 out:
 	if (moh) {

Modified: team/oej/pinequeue-1.8/apps/app_queue.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequeue-1.8/apps/app_queue.c?view=diff&rev=369920&r1=369919&r2=369920
==============================================================================
--- team/oej/pinequeue-1.8/apps/app_queue.c (original)
+++ team/oej/pinequeue-1.8/apps/app_queue.c Wed Jul 11 07:05:12 2012
@@ -4680,11 +4680,6 @@
 			int res2;
 
 			res2 = ast_autoservice_start(qe->chan);
-
-			/* instead of starting autoservice and jacking this thread to push sound to the
-			   peer channel, let's set up a background player to the peer channel and 
-			   get on with life in this thread. */
-			
 			if (qe->parent->memberdelay) {
 				ast_log(LOG_NOTICE, "Delaying member connect for %d seconds\n", qe->parent->memberdelay);
 				res2 |= ast_safe_sleep(peer, qe->parent->memberdelay * 1000);
@@ -4716,8 +4711,8 @@
 						}
 					}
 				}
-			}
-			res2 |= ast_autoservice_stop(qe->chan);
+				ast_autoservice_stop(qe->chan);
+			}
 
 
 			if (ast_check_hangup(peer)) {
@@ -4737,8 +4732,8 @@
 				ast_hangup(peer);
 				ao2_ref(member, -1);
 				goto out;
-			} else if (res2) {
-				/* Caller must have hung up just before being connected*/
+			} else if (ast_check_hangup(qe->chan)) {
+				/* Caller must have hung up just before being connected */
 				ast_log(LOG_NOTICE, "Caller was about to talk to agent on %s but the caller hungup.\n", peer->name);
 				ast_queue_log(queuename, qe->chan->uniqueid, member->membername, "ABANDON", "%d|%d|%ld", qe->pos, qe->opos, (long) time(NULL) - qe->start);
 				record_abandoned(qe);

Modified: team/oej/pinequeue-1.8/apps/app_stack.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequeue-1.8/apps/app_stack.c?view=diff&rev=369920&r1=369919&r2=369920
==============================================================================
--- team/oej/pinequeue-1.8/apps/app_stack.c (original)
+++ team/oej/pinequeue-1.8/apps/app_stack.c Wed Jul 11 07:05:12 2012
@@ -73,10 +73,14 @@
 			<parameter name="condition" required="true" />
 			<parameter name="destination" required="true" argsep=":">
 				<argument name="labeliftrue" hasparams="optional">
+					<para>Continue at <replaceable>labeliftrue</replaceable> if the condition is true.
+					Takes the form similar to Goto() of [[context,]extension,]priority.</para>
 					<argument name="arg1" required="true" multiple="true" />
 					<argument name="argN" />
 				</argument>
 				<argument name="labeliffalse" hasparams="optional">
+					<para>Continue at <replaceable>labeliffalse</replaceable> if the condition is false.
+					Takes the form similar to Goto() of [[context,]extension,]priority.</para>
 					<argument name="arg1" required="true" multiple="true" />
 					<argument name="argN" />
 				</argument>
@@ -93,6 +97,7 @@
 			<ref type="application">MacroIf</ref>
 			<ref type="function">IF</ref>
 			<ref type="application">GotoIf</ref>
+			<ref type="application">Goto</ref>
 		</see-also>
 	</application>
 	<application name="Return" language="en_US">

Modified: team/oej/pinequeue-1.8/apps/app_voicemail.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequeue-1.8/apps/app_voicemail.c?view=diff&rev=369920&r1=369919&r2=369920
==============================================================================
--- team/oej/pinequeue-1.8/apps/app_voicemail.c (original)
+++ team/oej/pinequeue-1.8/apps/app_voicemail.c Wed Jul 11 07:05:12 2012
@@ -1758,24 +1758,27 @@
 static int vm_allocate_dh(struct vm_state *vms, struct ast_vm_user *vmu, int count_msg) {
 
 	int arraysize = (vmu->maxmsg > count_msg ? vmu->maxmsg : count_msg);
-	if (!vms->dh_arraysize) {
-		/* initial allocation */
+
+	/* remove old allocation */
+	if (vms->deleted) {
+		ast_free(vms->deleted);
+		vms->deleted = NULL;
+	}
+	if (vms->heard) {
+		ast_free(vms->heard);
+		vms->heard = NULL;
+	}
+	vms->dh_arraysize = 0;
+
+	if (arraysize > 0) {
 		if (!(vms->deleted = ast_calloc(arraysize, sizeof(int)))) {
 			return -1;
 		}
 		if (!(vms->heard = ast_calloc(arraysize, sizeof(int)))) {
+			ast_free(vms->deleted);
+			vms->deleted = NULL;
 			return -1;
 		}
-		vms->dh_arraysize = arraysize;
-	} else if (vms->dh_arraysize < arraysize) {
-		if (!(vms->deleted = ast_realloc(vms->deleted, arraysize * sizeof(int)))) {
-			return -1;
-		}
-		if (!(vms->heard = ast_realloc(vms->heard, arraysize * sizeof(int)))) {
-			return -1;
-		}
-		memset(vms->deleted, 0, arraysize * sizeof(int));
-		memset(vms->heard, 0, arraysize * sizeof(int));
 		vms->dh_arraysize = arraysize;
 	}
 
@@ -6567,9 +6570,15 @@
 		name = "Unknown Caller";
 
 	/* If deleted, show "undeleted" */
-
-	if (vms->deleted[vms->curmsg])
-		keys[1] = ADSI_KEY_SKT | (ADSI_KEY_APPS + 11);
+#ifdef IMAP_STORAGE
+	ast_mutex_lock(&vms->lock);
+#endif
+	if (vms->deleted[vms->curmsg]) {
+ 		keys[1] = ADSI_KEY_SKT | (ADSI_KEY_APPS + 11);
+	}
+#ifdef IMAP_STORAGE
+	ast_mutex_unlock(&vms->lock);
+#endif
 
 	/* Except "Exit" */
 	keys[5] = ADSI_KEY_SKT | (ADSI_KEY_APPS + 5);
@@ -6622,8 +6631,15 @@
 	}
 
 	/* If deleted, show "undeleted" */
-	if (vms->deleted[vms->curmsg]) 
+#ifdef IMAP_STORAGE
+	ast_mutex_lock(&vms->lock);
+#endif
+	if (vms->deleted[vms->curmsg]) {
 		keys[1] = ADSI_KEY_SKT | (ADSI_KEY_APPS + 11);
+	}
+#ifdef IMAP_STORAGE
+	ast_mutex_unlock(&vms->lock);
+#endif
 
 	/* Except "Exit" */
 	keys[5] = ADSI_KEY_SKT | (ADSI_KEY_APPS + 5);
@@ -7805,8 +7821,12 @@
 
 	if (!res) {
 		make_file(vms->fn, sizeof(vms->fn), vms->curdir, vms->curmsg);
+#ifdef IMAP_STORAGE
+		ast_mutex_lock(&vms->lock);
+#endif
 		vms->heard[vms->curmsg] = 1;
 #ifdef IMAP_STORAGE
+		ast_mutex_unlock(&vms->lock);
 		/*IMAP storage stores any prepended message from a forward
 		 * as a separate file from the rest of the message
 		 */
@@ -8021,6 +8041,7 @@
 	}
 	ast_unlock_path(vms->curdir);
 #else /* defined(IMAP_STORAGE) */
+	ast_mutex_lock(&vms->lock);
 	if (vms->deleted) {
 		/* Since we now expunge after each delete, deleting in reverse order
 		 * ensures that no reordering occurs between each step. */
@@ -8035,12 +8056,18 @@
 #endif
 
 done:
-	if (vms->deleted && last_msg_idx) {
+	if (vms->deleted) {
 		ast_free(vms->deleted);
-	}
-	if (vms->heard && last_msg_idx) {
+		vms->deleted = NULL;
+	}
+	if (vms->heard) {
 		ast_free(vms->heard);
-	}
+		vms->heard = NULL;
+	}
+	vms->dh_arraysize = 0;
+#ifdef IMAP_STORAGE
+	ast_mutex_unlock(&vms->lock);
+#endif
 
 	return 0;
 }
@@ -9135,14 +9162,25 @@
 				res = ast_play_and_wait(chan, "vm-next");
 			}
 			if (!res) {
-				if (!vms->deleted[vms->curmsg])
+				int curmsg_deleted;
+#ifdef IMAP_STORAGE
+				ast_mutex_lock(&vms->lock);
+#endif
+				curmsg_deleted = vms->deleted[vms->curmsg];
+#ifdef IMAP_STORAGE
+				ast_mutex_unlock(&vms->lock);
+#endif
+				if (!curmsg_deleted) {
 					res = ast_play_and_wait(chan, "vm-delete");
-				else
+				} else {
 					res = ast_play_and_wait(chan, "vm-undelete");
-				if (!res)
+				}
+				if (!res) {
 					res = ast_play_and_wait(chan, "vm-toforward");
-				if (!res)
+				}
+				if (!res) {
 					res = ast_play_and_wait(chan, "vm-savemessage");
+				}
 			}
 		}
 		if (!res) {
@@ -10179,6 +10217,7 @@
 			}
 
 			vms.starting = 1;
+			vms.curmsg = 0;
 			break;
 		case '3': /* Advanced options */
 			ast_test_suite_event_notify("ADVOPTIONS", "Message: entering advanced options menu");

Modified: team/oej/pinequeue-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequeue-1.8/channels/chan_sip.c?view=diff&rev=369920&r1=369919&r2=369920
==============================================================================
--- team/oej/pinequeue-1.8/channels/chan_sip.c (original)
+++ team/oej/pinequeue-1.8/channels/chan_sip.c Wed Jul 11 07:05:12 2012
@@ -1232,7 +1232,7 @@
 static int handle_request_do(struct sip_request *req, struct ast_sockaddr *addr);
 static int sip_standard_port(enum sip_transport type, int port);
 static int sip_prepare_socket(struct sip_pvt *p);
-static int get_address_family_filter(const struct ast_sockaddr *addr);
+static int get_address_family_filter(unsigned int transport);
 
 /*--- Transmitting responses and requests */
 static int sipsock_read(int *id, int fd, short events, void *ignore);
@@ -1415,6 +1415,8 @@
 				      const char *name, int flag, int family);
 static int ast_sockaddr_resolve_first(struct ast_sockaddr *addr,
 				      const char *name, int flag);
+static int ast_sockaddr_resolve_first_transport(struct ast_sockaddr *addr,
+						const char *name, int flag, unsigned int transport);
 
 /*--- Debugging
 	Functions for enabling debug per IP or fully, or enabling history logging for
@@ -1516,7 +1518,7 @@
 static void build_callid_pvt(struct sip_pvt *pvt);
 static void change_callid_pvt(struct sip_pvt *pvt, const char *callid);
 static void build_callid_registry(struct sip_registry *reg, const struct ast_sockaddr *ourip, const char *fromdomain);
-static void make_our_tag(char *tagbuf, size_t len);
+static void make_our_tag(struct sip_pvt *pvt);
 static int add_header(struct sip_request *req, const char *var, const char *value);
 static int add_header_max_forwards(struct sip_pvt *dialog, struct sip_request *req);
 static int add_content(struct sip_request *req, const char *line);
@@ -3095,7 +3097,7 @@
 	if (!ast_sockaddr_parse(&proxy->ip, proxy->name, 0)) {
 		/* Ok, not an IP address, then let's check if it's a domain or host */
 		/* XXX Todo - if we have proxy port, don't do SRV */
-		proxy->ip.ss.ss_family = get_address_family_filter(&bindaddr); /* Filter address family */
+		proxy->ip.ss.ss_family = get_address_family_filter(SIP_TRANSPORT_UDP); /* Filter address family */
 		if (ast_get_ip_or_srv(&proxy->ip, proxy->name, sip_cfg.srvlookup ? "_sip._udp" : NULL) < 0) {
 				ast_log(LOG_WARNING, "Unable to locate host '%s'\n", proxy->name);
 				return FALSE;
@@ -3848,7 +3850,7 @@
 			ast_debug(3, "Re-scheduled destruction of SIP call %s\n", p->callid ? p->callid : "<unknown>");
 			append_history(p, "ReliableXmit", "timeout");
 			if (sscanf(p->lastmsg, "Tx: %30s", method_str) == 1 || sscanf(p->lastmsg, "Rx: %30s", method_str) == 1) {
-				if (method_match(SIP_CANCEL, method_str) || method_match(SIP_BYE, method_str)) {
+				if (p->ongoing_reinvite || method_match(SIP_CANCEL, method_str) || method_match(SIP_BYE, method_str)) {
 					pvt_set_needdestroy(p, "autodestruct");
 				}
 			}
@@ -4810,7 +4812,7 @@
 					if (ast_sockaddr_resolve(&addrs,
 								 tmp->value,
 								 PARSE_PORT_FORBID,
-								 get_address_family_filter(&bindaddr)) <= 0 ||
+								 get_address_family_filter(SIP_TRANSPORT_UDP)) <= 0 ||
 								 ast_sockaddr_cmp(&addrs[0], addr)) {
 						/* No match */
 						ast_variables_destroy(*var);
@@ -5478,7 +5480,7 @@
 			}
 		}
 
-		if (ast_sockaddr_resolve_first(&dialog->sa, hostn, 0)) {
+		if (ast_sockaddr_resolve_first_transport(&dialog->sa, hostn, 0, dialog->socket.type ? dialog->socket.type : SIP_TRANSPORT_UDP)) {
 			ast_log(LOG_WARNING, "No such host: %s\n", peername);
 			return -1;
 		}
@@ -6223,6 +6225,21 @@
 	return 0;
 }
 
+static int reinvite_timeout(const void *data)
+{
+	struct sip_pvt *dialog = (struct sip_pvt *) data;
+	struct ast_channel *owner = sip_pvt_lock_full(dialog);
+	dialog->reinviteid = -1;
+	check_pendings(dialog);
+	if (owner) {
+		ast_channel_unlock(owner);
+		ast_channel_unref(owner);
+	}
+	ao2_unlock(dialog);
+	dialog_unref(dialog, "unref for reinvite timeout");
+	return 0;
+}
+
 /*! \brief  sip_hangup: Hangup SIP call
  * Part of PBX interface, called from ast_hangup */
 static int sip_hangup(struct ast_channel *ast)
@@ -6412,8 +6429,16 @@
 				ast_set_flag(&p->flags[0], SIP_PENDINGBYE);	
 				ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);	
 				AST_SCHED_DEL_UNREF(sched, p->waitid, dialog_unref(p, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));
-				if (sip_cancel_destroy(p))
+				if (sip_cancel_destroy(p)) {
 					ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
+				}
+				/* If we have an ongoing reinvite, there is a chance that we have gotten a provisional
+				 * response, but something weird has happened and we will never receive a final response.
+				 * So, just in case, check for pending actions after a bit of time to trigger the pending
+				 * bye that we are setting above */
+				if (p->ongoing_reinvite && p->reinviteid < 0) {
+					p->reinviteid = ast_sched_add(sched, 32 * p->timer_t1, reinvite_timeout, dialog_ref(p, "ref for reinvite_timeout"));
+				}
 			}
 		}
 	}
@@ -6980,6 +7005,9 @@
 		}
 		break;
 	case AST_CONTROL_UPDATE_RTP_PEER: /* Absorb this since it is handled by the bridge */
+		break;
+	case AST_CONTROL_FLASH: /* We don't currently handle AST_CONTROL_FLASH here, but it is expected, so we don't need to warn either. */
+		res = -1;
 		break;
 	case -1:
 		res = -1;
@@ -7605,9 +7633,9 @@
 }
 
 /*! \brief Make our SIP dialog tag */
-static void make_our_tag(char *tagbuf, size_t len)
-{
-	snprintf(tagbuf, len, "as%08lx", ast_random());
+static void make_our_tag(struct sip_pvt *pvt)
+{
+	ast_string_field_build(pvt, tag, "as%08lx", ast_random());
 }
 
 /*! \brief Allocate Session-Timers struct w/in dialog */
@@ -7686,6 +7714,7 @@
 	p->method = intended_method;
 	p->initid = -1;
 	p->waitid = -1;
+	p->reinviteid = -1;
 	p->autokillid = -1;
 	p->request_queue_sched_id = -1;
 	p->provisional_keepalive_sched_id = -1;
@@ -7718,7 +7747,7 @@
 	p->do_history = recordhistory;
 
 	p->branch = ast_random();	
-	make_our_tag(p->tag, sizeof(p->tag));
+	make_our_tag(p);
 	p->ocseq = INITIAL_CSEQ;
 	p->allowed_methods = UINT_MAX;
 
@@ -7804,7 +7833,7 @@
 	}
 
 	if (via->maddr) {
-		if (ast_sockaddr_resolve_first(&p->sa, via->maddr, PARSE_PORT_FORBID)) {
+		if (ast_sockaddr_resolve_first_transport(&p->sa, via->maddr, PARSE_PORT_FORBID, p->socket.type)) {
 			ast_log(LOG_WARNING, "Can't find address for maddr '%s'\n", via->maddr);
 			ast_log(LOG_ERROR, "error processing via header\n");
 			free_via(via);
@@ -10092,7 +10121,7 @@
 
 	/*! \todo XXX If we have sip_cfg.srvlookup on, then look for NAPTR/SRV,
 	 * otherwise, just look for A records */
-	if (ast_sockaddr_resolve_first(&p->sa, hostname, 0)) {
+	if (ast_sockaddr_resolve_first_transport(&p->sa, hostname, 0, p->socket.type)) {
 		ast_log(LOG_WARNING, "Can't find address for host '%s'\n", hostname);
 		return;
 	}
@@ -10113,7 +10142,7 @@
 
 		/*! \todo XXX If we have sip_cfg.srvlookup on, then look for
 		 * NAPTR/SRV, otherwise, just look for A records */
-		if (ast_sockaddr_resolve_first(&p->sa, hostname, PARSE_PORT_FORBID)) {
+		if (ast_sockaddr_resolve_first_transport(&p->sa, hostname, PARSE_PORT_FORBID, p->socket.type)) {
 			ast_log(LOG_WARNING, "Can't find address for host '%s'\n", hostname);
 			return;
 		}
@@ -10584,7 +10613,7 @@
 	}
 
 	p->branch = ast_random();
-	make_our_tag(p->tag, sizeof(p->tag));
+	make_our_tag(p);
 	p->ocseq = INITIAL_CSEQ;
 
 	if (useglobal_nat && addr) {
@@ -11889,7 +11918,7 @@
 	initialize_initreq(p, &req);
 	p->lastinvite = p->ocseq;
 	ast_set_flag(&p->flags[0], SIP_OUTGOING);       /* Change direction of this dialog */
-
+	p->ongoing_reinvite = 1;
 	return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
 }
 
@@ -12451,7 +12480,7 @@
 		struct sip_subscription_mwi *saved;
 		snprintf(transport, sizeof(transport), "_%s._%s", get_srv_service(mwi->transport), get_srv_protocol(mwi->transport));
 
-		mwi->us.ss.ss_family = get_address_family_filter(&bindaddr); /* Filter address family */
+		mwi->us.ss.ss_family = get_address_family_filter(mwi->transport); /* Filter address family */
 		saved = ASTOBJ_REF(mwi);
 		ast_dnsmgr_lookup_cb(mwi->hostname, &mwi->us, &mwi->dnsmgr, sip_cfg.srvlookup ? transport : NULL, on_dns_update_mwi, saved);
 		if (!mwi->dnsmgr) {
@@ -13242,7 +13271,7 @@
 		char transport[MAXHOSTNAMELEN];
 		peer = find_peer(r->hostname, NULL, TRUE, FINDPEERS, FALSE, 0);
 		snprintf(transport, sizeof(transport), "_%s._%s",get_srv_service(r->transport), get_srv_protocol(r->transport)); /* have to use static get_transport function */
-		r->us.ss.ss_family = get_address_family_filter(&bindaddr); /* Filter address family */
+		r->us.ss.ss_family = get_address_family_filter(r->transport); /* Filter address family */
 
 		/* No point in doing a DNS lookup of the register hostname if we're just going to
 		 * end up using an outbound proxy. obproxy_get is safe to call with either of r->call
@@ -13267,7 +13296,7 @@
 			return 0;
 		} else {
 			p = dialog_ref(r->call, "getting a copy of the r->call dialog in transmit_register");
-			make_our_tag(p->tag, sizeof(p->tag));	/* create a new local tag for every register attempt */
+			make_our_tag(p);	/* create a new local tag for every register attempt */
 			ast_string_field_set(p, theirtag, NULL);	/* forget their old tag, so we don't match tags when getting response */
 		}
 	} else {
@@ -13958,7 +13987,7 @@
 		return -1;
 	}
 
-	if (ast_sockaddr_resolve_first(addr, hostport, 0)) {
+	if (ast_sockaddr_resolve_first_transport(addr, hostport, 0, get_transport_str2enum(transport))) {
 		ast_log(LOG_WARNING, "Invalid host name in Contact: (can't "
 			"resolve in DNS) : '%s'\n", hostport);
 		return -1;
@@ -14103,7 +14132,7 @@
 		ast_debug(1, "Store REGISTER's Contact header for call routing.\n");
 		/* XXX This could block for a long time XXX */
 		/*! \todo Check NAPTR/SRV if we have not got a port in the URI */
-		if (ast_sockaddr_resolve_first(&testsa, hostport, 0)) {
+		if (ast_sockaddr_resolve_first_transport(&testsa, hostport, 0, peer->socket.type)) {
 			ast_log(LOG_WARNING, "Invalid hostport '%s'\n", hostport);
 			ast_string_field_set(peer, fullcontact, "");
 			ast_string_field_set(pvt, our_contact, "");
@@ -18988,7 +19017,7 @@
 /*! \brief Enable SIP Debugging for a single IP */
 static char *sip_do_debug_ip(int fd, const char *arg)
 {
-	if (ast_sockaddr_resolve_first(&debugaddr, arg, 0)) {
+	if (ast_sockaddr_resolve_first_af(&debugaddr, arg, 0, 0)) {
 		return CLI_SHOWUSAGE;
 	}
 
@@ -19838,8 +19867,11 @@
 static void check_pendings(struct sip_pvt *p)
 {
 	if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
-		/* if we can't BYE, then this is really a pending CANCEL */
-		if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA) {
+		if (p->reinviteid > -1) {
+			/* Outstanding p->reinviteid timeout, so wait... */
+			return;
+		} else if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA) {
+			/* if we can't BYE, then this is really a pending CANCEL */
 			p->invitestate = INV_CANCELLED;
 			transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
 			/* If the cancel occurred on an initial invite, cancel the pending BYE */
@@ -19850,8 +19882,9 @@
 			   INVITE, but do set an autodestruct just in case we never get it. */
 		} else {
 			/* We have a pending outbound invite, don't send something
-				new in-transaction */
-			if (p->pendinginvite)
+			 * new in-transaction, unless it is a pending reinvite, then
+			 * by the time we are called here, we should probably just hang up. */
+			if (p->pendinginvite && !p->ongoing_reinvite)
 				return;
 
 			if (p->owner) {
@@ -20091,9 +20124,17 @@
  	if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
  		p->invitestate = INV_COMPLETED;
  	
+	if ((resp >= 200 && reinvite)) {
+		p->ongoing_reinvite = 0;
+		if (p->reinviteid > -1) {
+			AST_SCHED_DEL_UNREF(sched, p->reinviteid, dialog_unref(p, "unref dialog for reinvite timeout because of a final response"));
+		}
+	}
+
 	/* Final response, clear out pending invite */
-	if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite)
+	if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite) {
 		p->pendinginvite = 0;
+	}
 
 	/* If this is a response to our initial INVITE, we need to set what we can use
 	 * for this peer.
@@ -22777,8 +22818,6 @@
 				 */
 				/* Fall through */
 			case SIP_GET_DEST_EXTEN_NOT_FOUND:
-			case SIP_GET_DEST_REFUSED:
-			default:
 				{
 					char *decoded_exten = ast_strdupa(p->exten);
 					transmit_response_reliable(p, "404 Not Found", req);
@@ -22787,6 +22826,10 @@
 						" '%s' rejected because extension not found in context '%s'.\n",
 						S_OR(p->username, p->peername), ast_sockaddr_stringify(&p->recv), decoded_exten, p->context);
 				}
+				break;
+			case SIP_GET_DEST_REFUSED:
+			default:
+				transmit_response_reliable(p, "403 Forbidden", req);
 			} /* end switch */
 
 			p->invitestate = INV_COMPLETED;
@@ -22802,7 +22845,7 @@
 				ast_string_field_set(p, exten, "s");
 			/* Initialize our tag */
 
-			make_our_tag(p->tag, sizeof(p->tag));
+			make_our_tag(p);
 			/* First invitation - create the channel.  Allocation
 			 * failures are handled below. */
 			c = sip_new(p, AST_STATE_DOWN, S_OR(p->peername, NULL), NULL);
@@ -24774,7 +24817,7 @@
 
 	/* Initialize tag for new subscriptions */	
 	if (ast_strlen_zero(p->tag))
-		make_our_tag(p->tag, sizeof(p->tag));
+		make_our_tag(p);
 
 	if (!strncmp(eventheader, "presence", MAX(event_len, 8)) || !strncmp(eventheader, "dialog", MAX(event_len, 6))) { /* Presence, RFC 3842 */
 		unsigned int pidf_xml;
@@ -25266,14 +25309,15 @@
 		if (!p->initreq.headers && req->has_to_tag) {
 			/* If this is a first request and it got a to-tag, it is not for us */
 			if (!req->ignore && req->method == SIP_INVITE) {
-				/* We will be subversive here. By blanking out the to-tag of the request,
-				 * it will cause us to attach our own generated to-tag instead. This way,
-				 * when we receive an ACK, the ACK will contain the to-tag we generated,
-				 * resulting in a proper to-tag match.
+				/* Just because we think this is a dialog-starting INVITE with a to-tag
+				 * doesn't mean it actually is. It could be a reinvite for an established, but
+				 * unknown dialog. In such a case, we need to change our tag to the
+				 * incoming INVITE's to-tag so that they will recognize the 481 we send and
+				 * so that we will properly match their incoming ACK.
 				 */
-				char *to_header = (char *) get_header(req, "To");
-				char *tag = strstr(to_header, ";tag=");
-				*tag = '\0';
+				char totag[128];
+				gettag(req, "To", totag, sizeof(totag));
+				ast_string_field_set(p, tag, totag);
 				p->pendinginvite = p->icseq;
 				transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
 				/* Will cease to exist after ACK */
@@ -25551,8 +25595,20 @@
  *
  * \note return 0 if addr is [::] else it returns addr's family.
  */
-int get_address_family_filter(const struct ast_sockaddr *addr)
-{
+int get_address_family_filter(unsigned int transport)
+{
+	const struct ast_sockaddr *addr = NULL;
+
+	if ((transport == SIP_TRANSPORT_UDP) || !transport) {
+		addr = &bindaddr;
+	}
+	else if (transport == SIP_TRANSPORT_TCP) {
+		addr = &sip_tcp_desc.local_address;
+	}
+	else if (transport == SIP_TRANSPORT_TLS) {
+		addr = &sip_tls_desc.local_address;
+	}
+
 	if (ast_sockaddr_is_ipv6(addr) && ast_sockaddr_is_any(addr)) {
 		return 0;
 	}
@@ -26687,7 +26743,7 @@
 	}
 
 	if (!ast_strlen_zero(remote_address)) {
-		if (ast_sockaddr_resolve_first(&remote_address_sa, remote_address, 0)) {
+		if (ast_sockaddr_resolve_first_transport(&remote_address_sa, remote_address, 0, transport)) {
 			ast_log(LOG_WARNING, "Unable to find IP address for host %s. We will not use this remote IP address\n", remote_address);
 		} else {
 			if (!ast_sockaddr_port(&remote_address_sa)) {
@@ -27909,7 +27965,7 @@
 
 		snprintf(transport, sizeof(transport), "_%s._%s", get_srv_service(peer->socket.type), get_srv_protocol(peer->socket.type));
 
-		peer->addr.ss.ss_family = get_address_family_filter(&bindaddr); /* Filter address family */
+		peer->addr.ss.ss_family = get_address_family_filter(peer->socket.type); /* Filter address family */
 		if (ast_dnsmgr_lookup_cb(_srvlookup, &peer->addr, &peer->dnsmgr, sip_cfg.srvlookup && !peer->portinuri ? transport : NULL,
 					on_dns_update_peer, ref_peer(peer, "Store peer on dnsmgr"))) {
 			ast_log(LOG_ERROR, "srvlookup failed for host: %s, on peer %s, removing peer\n", _srvlookup, peer->name);
@@ -29948,7 +30004,17 @@
 static int ast_sockaddr_resolve_first(struct ast_sockaddr *addr,
 				      const char* name, int flag)
 {
-	return ast_sockaddr_resolve_first_af(addr, name, flag, get_address_family_filter(&bindaddr));
+	return ast_sockaddr_resolve_first_af(addr, name, flag, get_address_family_filter(SIP_TRANSPORT_UDP));
+}
+
+/*! \brief  Return the first entry from ast_sockaddr_resolve filtered by family of binddaddr
+ *
+ * \warn Using this function probably means you have a faulty design.
+ */
+static int ast_sockaddr_resolve_first_transport(struct ast_sockaddr *addr,
+						const char* name, int flag, unsigned int transport)
+{
+        return ast_sockaddr_resolve_first_af(addr, name, flag, get_address_family_filter(transport));
 }
 
 /*! \brief

Modified: team/oej/pinequeue-1.8/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequeue-1.8/channels/sip/include/sip.h?view=diff&rev=369920&r1=369919&r2=369920
==============================================================================
--- team/oej/pinequeue-1.8/channels/sip/include/sip.h (original)
+++ team/oej/pinequeue-1.8/channels/sip/include/sip.h Wed Jul 11 07:05:12 2012
@@ -959,6 +959,7 @@
 		AST_STRING_FIELD(rdnis);        /*!< Referring DNIS */
 		AST_STRING_FIELD(redircause);   /*!< Referring cause */
 		AST_STRING_FIELD(theirtag);     /*!< Their tag */
+		AST_STRING_FIELD(tag);          /*!< Our tag for this session */
 		AST_STRING_FIELD(username);     /*!< [user] name */
 		AST_STRING_FIELD(peername);     /*!< [peer] name, not set if [user] */
 		AST_STRING_FIELD(authname);     /*!< Who we use for authentication */
@@ -1006,7 +1007,6 @@
 	                                       *   for incoming calls
 	                                       */
 	unsigned short req_secure_signaling:1;/*!< Whether we are required to have secure signaling or not */
-	char tag[11];                     /*!< Our tag for this session */
 	int timer_t1;                     /*!< SIP timer T1, ms rtt */
 	int timer_b;                      /*!< SIP timer B, ms */
 	unsigned int sipoptions;          /*!< Supported SIP options on the other end */
@@ -1056,6 +1056,7 @@
 	struct sip_auth_container *peerauth;/*!< Realm authentication credentials */
 	int noncecount;                     /*!< Nonce-count */
 	unsigned int stalenonce:1;          /*!< Marks the current nonce as responded too */
+	unsigned int ongoing_reinvite:1;    /*!< There is a reinvite in progress that might need to be cleaned up */
 	char lastmsg[256];                  /*!< Last Message sent/received */
 	int amaflags;                       /*!< AMA Flags */
 	uint32_t pendinginvite; /*!< Any pending INVITE or state NOTIFY (in subscribe pvt's) ? (seqno of this) */
@@ -1068,6 +1069,7 @@
 
 	int initid;                         /*!< Auto-congest ID if appropriate (scheduler) */
 	int waitid;                         /*!< Wait ID for scheduler after 491 or other delays */
+	int reinviteid;                     /*!< Reinvite in case of provisional, but no final response */
 	int autokillid;                     /*!< Auto-kill ID (scheduler) */
 	int t38id;                          /*!< T.38 Response ID */
 	struct sip_refer *refer;            /*!< REFER: SIP transfer data structure */

Modified: team/oej/pinequeue-1.8/configs/sip_notify.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequeue-1.8/configs/sip_notify.conf.sample?view=diff&rev=369920&r1=369919&r2=369920
==============================================================================
--- team/oej/pinequeue-1.8/configs/sip_notify.conf.sample (original)
+++ team/oej/pinequeue-1.8/configs/sip_notify.conf.sample Wed Jul 11 07:05:12 2012
@@ -16,6 +16,11 @@
 
 [aastra-xml]
 Event=>aastra-xml
+
+; Digium
+
+[digium-check-cfg]
+Event=>check-sync
 
 ; Linksys
 

Modified: team/oej/pinequeue-1.8/include/asterisk/adsi.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequeue-1.8/include/asterisk/adsi.h?view=diff&rev=369920&r1=369919&r2=369920
==============================================================================
--- team/oej/pinequeue-1.8/include/asterisk/adsi.h (original)
+++ team/oej/pinequeue-1.8/include/asterisk/adsi.h Wed Jul 11 07:05:12 2012
@@ -24,7 +24,6 @@
 #define _ASTERISK_ADSI_H
 
 #include "asterisk/callerid.h"
-#include "asterisk/optional_api.h"
 
 /*! \name ADSI parameters */
 /*@{ */
@@ -121,11 +120,13 @@
 #define ADSI_DIR_FROM_LEFT (0)
 #define ADSI_DIR_FROM_RIGHT (1)
 
+#define AST_ADSI_VERSION 1
+
 /*@} */
 
-AST_OPTIONAL_API(int, ast_adsi_begin_download, (struct ast_channel *chan, char *service, unsigned char *fdn, unsigned char *sec, int version), { return 0; });
-
-AST_OPTIONAL_API(int, ast_adsi_end_download, (struct ast_channel *chan), { return 0; });
+int ast_adsi_begin_download(struct ast_channel *chan, char *service, unsigned char *fdn, unsigned char *sec, int version);
+
+int ast_adsi_end_download(struct ast_channel *chan);
 
 /*! Restore ADSI initialization (for applications that play with ADSI
  *   and want to restore it to normal.  If you touch "INFO" then you
@@ -135,7 +136,7 @@
  * \retval 0 on success (or adsi unavailable)
  * \retval -1 on hangup
  */
-AST_OPTIONAL_API(int, ast_adsi_channel_restore, (struct ast_channel *chan), { return 0; });
+int ast_adsi_channel_restore(struct ast_channel *chan);
 
 /*!
  * \brief Display some stuff on the screen
@@ -147,7 +148,7 @@
  * \retval 0 on success (or adsi unavailable)
  * \retval -1 on hangup
  */
-AST_OPTIONAL_API(int, ast_adsi_print, (struct ast_channel *chan, char **lines, int *align, int voice), { return 0; });
+int ast_adsi_print(struct ast_channel *chan, char **lines, int *align, int voice);
 
 /*!
  * \brief Check if scripts for a given app are already loaded.
@@ -161,16 +162,16 @@
  * \retval -1 on hangup
  * \retval 1 if script already loaded.
  */
-AST_OPTIONAL_API(int, ast_adsi_load_session, (struct ast_channel *chan, unsigned char *app, int ver, int data), { return 0; });
-AST_OPTIONAL_API(int, ast_adsi_unload_session, (struct ast_channel *chan), { return 0; });
+int ast_adsi_load_session(struct ast_channel *chan, unsigned char *app, int ver, int data);
+int ast_adsi_unload_session(struct ast_channel *chan);
 
 /* ADSI Layer 2 transmission functions */
-AST_OPTIONAL_API(int, ast_adsi_transmit_message, (struct ast_channel *chan, unsigned char *msg, int msglen, int msgtype), { return 0; });
-AST_OPTIONAL_API(int, ast_adsi_transmit_message_full, (struct ast_channel *chan, unsigned char *msg, int msglen, int msgtype, int dowait), { return 0; });
+int ast_adsi_transmit_message(struct ast_channel *chan, unsigned char *msg, int msglen, int msgtype);
+int ast_adsi_transmit_message_full(struct ast_channel *chan, unsigned char *msg, int msglen, int msgtype, int dowait);
 /*! Read some encoded DTMF data.
  * Returns number of bytes received
  */
-AST_OPTIONAL_API(int, ast_adsi_read_encoded_dtmf, (struct ast_channel *chan, unsigned char *buf, int maxlen), { return 0; });
+int ast_adsi_read_encoded_dtmf(struct ast_channel *chan, unsigned char *buf, int maxlen);
 
 /* ADSI Layer 3 creation functions */
 
@@ -184,22 +185,22 @@
  * \retval -1 on error.
  */
 
-AST_OPTIONAL_API(int, ast_adsi_connect_session, (unsigned char *buf, unsigned char *fdn, int ver), { return 0; });
+int ast_adsi_connect_session(unsigned char *buf, unsigned char *fdn, int ver);
 
 /*! Build Query CPE ID of equipment.
  *  Returns number of bytes added to message
  */
-AST_OPTIONAL_API(int, ast_adsi_query_cpeid, (unsigned char *buf), { return 0; });
-AST_OPTIONAL_API(int, ast_adsi_query_cpeinfo, (unsigned char *buf), { return 0; });
+int ast_adsi_query_cpeid(unsigned char *buf);
+int ast_adsi_query_cpeinfo(unsigned char *buf);
 
 /*! Get CPE ID from an attached ADSI compatible CPE.
  * Returns 1 on success, storing 4 bytes of CPE ID at buf
  * or -1 on hangup, or 0 if there was no hangup but it failed to find the
  * device ID.  Returns to voice mode if "voice" is non-zero.
  */
-AST_OPTIONAL_API(int, ast_adsi_get_cpeid, (struct ast_channel *chan, unsigned char *cpeid, int voice), { return 0; });
-
-AST_OPTIONAL_API(int, ast_adsi_get_cpeinfo, (struct ast_channel *chan, int *width, int *height, int *buttons, int voice), { return 0; });
+int ast_adsi_get_cpeid(struct ast_channel *chan, unsigned char *cpeid, int voice);
+
+int ast_adsi_get_cpeinfo(struct ast_channel *chan, int *width, int *height, int *buttons, int voice);
 
 /*!
  * \brief Begin an ADSI script download
@@ -213,7 +214,7 @@
  * \retval -1 on error.
  */
 
-AST_OPTIONAL_API(int, ast_adsi_download_connect, (unsigned char *buf, char *service, unsigned char *fdn, unsigned char *sec, int ver), { return 0; });
+int ast_adsi_download_connect(unsigned char *buf, char *service, unsigned char *fdn, unsigned char *sec, int ver);
 
 /*!
  * \brief Disconnects a running session.
@@ -222,7 +223,7 @@
  * \retval number of bytes added to buffer
  * \retval -1 on error.
  */
-AST_OPTIONAL_API(int, ast_adsi_disconnect_session, (unsigned char *buf), { return 0; });
+int ast_adsi_disconnect_session(unsigned char *buf);
 
 /*!
  * \brief Disconnects (and hopefully saves) a downloaded script
@@ -231,7 +232,7 @@
  * \retval number of bytes added to buffer
  * \retval -1 on error.
  */
-AST_OPTIONAL_API(int, ast_adsi_download_disconnect, (unsigned char *buf), { return 0; });
+int ast_adsi_download_disconnect(unsigned char *buf);
 
 /*!
  * \brief Puts CPE in data mode.
@@ -240,9 +241,9 @@
  * \retval number of bytes added to buffer
  * \retval -1 on error.
  */
-AST_OPTIONAL_API(int, ast_adsi_data_mode, (unsigned char *buf), { return 0; });
-AST_OPTIONAL_API(int, ast_adsi_clear_soft_keys, (unsigned char *buf), { return 0; });
-AST_OPTIONAL_API(int, ast_adsi_clear_screen, (unsigned char *buf), { return 0; });
+int ast_adsi_data_mode(unsigned char *buf);
+int ast_adsi_clear_soft_keys(unsigned char *buf);
+int ast_adsi_clear_screen(unsigned char *buf);
 
 /*!
  * \brief Puts CPE in voice mode.
@@ -252,13 +253,13 @@
  * \retval number of bytes added to buffer
  * \retval -1 on error.
  */
-AST_OPTIONAL_API(int, ast_adsi_voice_mode, (unsigned char *buf, int when), { return 0; });
+int ast_adsi_voice_mode(unsigned char *buf, int when);
 
 /*!
  * \brief Returns non-zero if Channel does or might support ADSI
  * \param chan Channel to check
  */
-AST_OPTIONAL_API(int, ast_adsi_available, (struct ast_channel *chan), { return 0; });
+int ast_adsi_available(struct ast_channel *chan);
 
 /*!
  * \brief Loads a line of info into the display.
@@ -274,7 +275,7 @@
  * \retval -1 on error.
  */
 
-AST_OPTIONAL_API(int, ast_adsi_display, (unsigned char *buf, int page, int line, int just, int wrap, char *col1, char *col2), { return 0; });
+int ast_adsi_display(unsigned char *buf, int page, int line, int just, int wrap, char *col1, char *col2);
 
 /*!
  * \brief Sets the current line and page.
@@ -286,7 +287,7 @@
  * \retval -1 on error.
  */
 
-AST_OPTIONAL_API(int, ast_adsi_set_line, (unsigned char *buf, int page, int line), { return 0; });
+int ast_adsi_set_line(unsigned char *buf, int page, int line);
 
 /*!
  * \brief Creates "load soft key" parameters
@@ -300,7 +301,7 @@
  * \retval number of bytes added to buffer
  * \retval -1 on error.
  */
-AST_OPTIONAL_API(int, ast_adsi_load_soft_key, (unsigned char *buf, int key, const char *llabel, const char *slabel, char *ret, int data), { return 0; });
+int ast_adsi_load_soft_key(unsigned char *buf, int key, const char *llabel, const char *slabel, char *ret, int data);
 
 /*!
  * \brief Set which soft keys should be displayed
@@ -311,7 +312,7 @@
  * \retval number of bytes added to buffer
  * \retval -1 on error.
  */
-AST_OPTIONAL_API(int, ast_adsi_set_keys, (unsigned char *buf, unsigned char *keys), { return 0; });
+int ast_adsi_set_keys(unsigned char *buf, unsigned char *keys);
 
 /*!
  * \brief Set input information
@@ -325,7 +326,7 @@
  * \retval number of bytes added to buffer
  * \retval -1 on error.
  */
-AST_OPTIONAL_API(int, ast_adsi_input_control, (unsigned char *buf, int page, int line, int display, int format, int just), { return 0; });
+int ast_adsi_input_control(unsigned char *buf, int page, int line, int display, int format, int just);
 

[... 1161 lines stripped ...]



More information about the asterisk-commits mailing list