Restore SIP DTMF overlap dialing method.

The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
working correctly removed a long standing ability to do overlap dialing
using DTMF in the early media phase of a call.

See ASTERISK-18702 it has a very good description of the issue.

I started with Pavel Troller's chan_sip.diff patch on issue
ASTERISK-18702.

* Added 'dtmf' enum value to sip.conf allowoverlap config option.  The new
option value causes the Incomplte application to not send anything with
chan_sip so the caller can supply more digits via DTMF.

* Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means.

* Fixed get_destination() inconsistency with the pickup extension
matching.

* Fixed initialization of PAGE3 of global_flags in reload_config().

(closes issue ASTERISK-18702)
Reported by: Pavel Troller

Review: https://reviewboard.asterisk.org/r/1517/

Review: https://reviewboard.asterisk.org/r/1582/
........

Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345275 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345276 65c4cc65-6c06-0410-ace0-fbb531ad65f3

4 files changed, 146 insertions, 74 deletions

diff --git a/UPGRADE-1.8.txt b/UPGRADE-1.8.txt
index 75efabe7f..45edd0344 100644
--- a/UPGRADE-1.8.txt
+++ b/UPGRADE-1.8.txt
@@ -20,6 +20,10 @@
 
 From 1.6.2 to 1.8:
 
+* chan_sip no longer sets HASH(SIP_CAUSE,<chan name>) on channels by default.
+  This must now be enabled by setting 'sipstorecause' to 'yes' in sip.conf.
+  This carries a performance penalty.
+
 * Asterisk now requires libpri 1.4.11+ for PRI support.
 
 * A couple of CLI commands in res_ais were changed back to their original form:
@@ -92,8 +96,8 @@ From 1.6.2 to 1.8:
 * ExternalIVR will now send Z events for invalid or missing files, T events
   now include the interrupted file and bugs in argument parsing have been
   fixed so there may be arguments specified in incorrect ways that were
-  working that will no longer work.
-  Please see doc/externalivr.txt for details.
+  working that will no longer work. Please see 
+  https://wiki.asterisk.org/wiki/display/AST/External+IVR+Interface for details.
 
 * OSP lookup application changes following variable names:
   OSPPEERIP to OSPINPEERIP
@@ -155,6 +159,13 @@ From 1.6.2 to 1.8:
   changes to the files will not be detected.  You can revert to polling the
   directory by specifying --without-inotify to configure before compiling.
 
+* The 'sipusers' realtime table has been removed completely. Use the 'sippeers'
+  table with type 'user' for user type objects.
+
+* The sip.conf allowoverlap option now accepts 'dtmf' as a value.  If you
+  are using the early media DTMF overlap dialing method you now need to set
+  allowoverlap=dtmf.
+
 From 1.6.1 to 1.6.2:
 
 * SIP no longer sends the 183 progress message for early media by
@@ -250,6 +261,11 @@ From 1.6.1 to 1.6.2:
 * The cdr.conf file must exist and be configured correctly in order for CDR
   records to be written.
 
+* cdr_pgsql now assumes the encoding of strings it is handed are in LATIN9,
+  which should cover most uses of the extended ASCII set.  If your strings
+  use a different encoding in Asterisk, the "encoding" parameter may be set
+  to specify the correct character set.
+
 From 1.6.0.1 to 1.6.1:
 
 * The ast_agi_register_multiple() and ast_agi_unregister_multiple()
@@ -311,6 +327,3 @@ From 1.6.0.x to 1.6.1:
   which should be a char(8) or larger.  This field specifies whether or not a
   message has been designated to be "Urgent", "PRIORITY", or not.
 
-* The 'sipusers' realtime table has been removed completely. Use the 'sippeers'
-  table with type 'user' for user type objects.
-
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index fa6b6ea4c..d5401156a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1384,6 +1384,7 @@ static void  print_group(int fd, ast_group_t group, int crlf);
 static const char *dtmfmode2str(int mode) attribute_const;
 static int str2dtmfmode(const char *str) attribute_unused;
 static const char *insecure2str(int mode) attribute_const;
+static const char *allowoverlap2str(int mode) attribute_const;
 static void cleanup_stale_contexts(char *new, char *old);
 static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
 static const char *domain_mode_to_text(const enum domain_mode mode);
@@ -6837,17 +6838,25 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
 		break;
 	case AST_CONTROL_INCOMPLETE:
 		if (ast->_state != AST_STATE_UP) {
-			if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
+			switch (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
+			case SIP_PAGE2_ALLOWOVERLAP_YES:
 				transmit_response_reliable(p, "484 Address Incomplete", &p->initreq);
-			} else {
+				p->invitestate = INV_COMPLETED;
+				sip_alreadygone(p);
+				ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
+				break;
+			case SIP_PAGE2_ALLOWOVERLAP_DTMF:
+				/* Just wait for inband DTMF digits */
+				break;
+			default:
+				/* it actually means no support for overlap */
 				transmit_response_reliable(p, "404 Not Found", &p->initreq);
+				p->invitestate = INV_COMPLETED;
+				sip_alreadygone(p);
+				ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
+				break;
 			}
-			p->invitestate = INV_COMPLETED;
-			sip_alreadygone(p);
-			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
-			break;
 		}
-		res = 0;
 		break;
 	case AST_CONTROL_PROCEEDING:
 		if ((ast->_state != AST_STATE_UP) &&
@@ -15506,18 +15515,23 @@ static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_re
 		}
 	} else {
 		struct ast_cc_agent *agent;
-		int which = 0;
 		/* Check the dialplan for the username part of the request URI,
 		   the domain will be stored in the SIPDOMAIN variable
 		   Return 0 if we have a matching extension */
-		if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from)) ||
-		    (ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from)) && (which = 1)) ||
-		    !strcmp(decoded_uri, ast_pickup_ext())) {
+		if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from))) {
 			if (!oreq) {
-				ast_string_field_set(p, exten, which ? decoded_uri : uri);
+				ast_string_field_set(p, exten, uri);
 			}
 			return SIP_GET_DEST_EXTEN_FOUND;
-		} else if ((agent = find_sip_cc_agent_by_notify_uri(tmp))) {
+		}
+		if (ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))
+			|| !strcmp(decoded_uri, ast_pickup_ext())) {
+			if (!oreq) {
+				ast_string_field_set(p, exten, decoded_uri);
+			}
+			return SIP_GET_DEST_EXTEN_FOUND;
+		}
+		if ((agent = find_sip_cc_agent_by_notify_uri(tmp))) {
 			struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
 			/* This is a CC recall. We can set p's extension to the exten from
 			 * the original INVITE
@@ -15536,11 +15550,12 @@ static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_re
 		}
 	}
 
-	/* Return 1 for pickup extension or overlap dialling support (if we support it) */
-	if((ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) &&
- 	    ast_canmatch_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))) ||
-	    !strncmp(decoded_uri, ast_pickup_ext(), strlen(decoded_uri))) {
-		return SIP_GET_DEST_PICKUP_EXTEN_FOUND;
+	if (ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)
+		&& (ast_canmatch_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from))
+			|| ast_canmatch_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))
+			|| !strncmp(decoded_uri, ast_pickup_ext(), strlen(decoded_uri)))) {
+		/* Overlap dialing is enabled and we need more digits to match an extension. */
+		return SIP_GET_DEST_EXTEN_MATCHMORE;
 	}
 
 	return SIP_GET_DEST_EXTEN_NOT_FOUND;
@@ -17150,6 +17165,19 @@ static const char *insecure2str(int mode)
 	return map_x_s(insecurestr, mode, "<error>");
 }
 
+static const struct _map_x_s allowoverlapstr[] = {
+	{ SIP_PAGE2_ALLOWOVERLAP_YES,   "Yes" },
+	{ SIP_PAGE2_ALLOWOVERLAP_DTMF,  "DTMF" },
+	{ SIP_PAGE2_ALLOWOVERLAP_NO,    "No" },
+	{ -1,                           NULL }, /* terminator */
+};
+
+/*! \brief Convert AllowOverlap setting to printable string */
+static const char *allowoverlap2str(int mode)
+{
+	return map_x_s(allowoverlapstr, mode, "<error>");
+}
+
 /*! \brief Destroy disused contexts between reloads
 	Only used in reload_config so the code for regcontext doesn't get ugly
 */
@@ -17696,7 +17724,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
 		ast_cli(fd, "  Trust RPID   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_TRUSTRPID)));
 		ast_cli(fd, "  Send RPID    : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_SENDRPID)));
 		ast_cli(fd, "  Subscriptions: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
-		ast_cli(fd, "  Overlap dial : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
+		ast_cli(fd, "  Overlap dial : %s\n", allowoverlap2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
 		if (peer->outboundproxy)
 			ast_cli(fd, "  Outb. proxy  : %s %s\n", ast_strlen_zero(peer->outboundproxy->name) ? "<not set>" : peer->outboundproxy->name,
 							peer->outboundproxy->force ? "(forced)" : "");
@@ -18253,7 +18281,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
 	ast_cli(a->fd, "  Match Auth Username:    %s\n", AST_CLI_YESNO(global_match_auth_username));
 	ast_cli(a->fd, "  Allow unknown access:   %s\n", AST_CLI_YESNO(sip_cfg.allowguest));
 	ast_cli(a->fd, "  Allow subscriptions:    %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
-	ast_cli(a->fd, "  Allow overlap dialing:  %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
+	ast_cli(a->fd, "  Allow overlap dialing:  %s\n", allowoverlap2str(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
 	ast_cli(a->fd, "  Allow promisc. redir:   %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_PROMISCREDIR)));
 	ast_cli(a->fd, "  Enable call counters:   %s\n", AST_CLI_YESNO(global_callcounter));
 	ast_cli(a->fd, "  SIP domain support:     %s\n", AST_CLI_YESNO(!AST_LIST_EMPTY(&domain_list)));
@@ -21546,10 +21574,13 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
 					break;
 				case 484: /* Address Incomplete */
 					if (owner && sipmethod != SIP_BYE) {
-						if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
+						switch (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
+						case SIP_PAGE2_ALLOWOVERLAP_YES:
 							ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
-						} else {
+							break;
+						default:
 							ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(404));
+							break;
 						}
 					}
 					break;
@@ -22262,7 +22293,7 @@ static int handle_request_options(struct sip_pvt *p, struct sip_request *req, st
 		case SIP_GET_DEST_INVALID_URI:
 			msg = "416 Unsupported URI scheme";
 			break;
-		case SIP_GET_DEST_PICKUP_EXTEN_FOUND:
+		case SIP_GET_DEST_EXTEN_MATCHMORE:
 		case SIP_GET_DEST_REFUSED:
 		case SIP_GET_DEST_EXTEN_NOT_FOUND:
 			//msg = "404 Not Found";
@@ -22983,12 +23014,21 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 			case SIP_GET_DEST_INVALID_URI:
 				transmit_response_reliable(p, "416 Unsupported URI scheme", req);
 				break;
-			case SIP_GET_DEST_PICKUP_EXTEN_FOUND:
-				if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
+			case SIP_GET_DEST_EXTEN_MATCHMORE:
+				if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)
+					== SIP_PAGE2_ALLOWOVERLAP_YES) {
 					transmit_response_reliable(p, "484 Address Incomplete", req);
 					break;
 				}
-			/* INTENTIONAL FALL THROUGH */
+				/*
+				 * XXX We would have to implement collecting more digits in
+				 * chan_sip for any other schemes of overlap dialing.
+				 *
+				 * For SIP_PAGE2_ALLOWOVERLAP_DTMF it is better to do this in
+				 * the dialplan using the Incomplete application rather than
+				 * having the channel driver do it.
+				 */
+				/* Fall through */
 			case SIP_GET_DEST_EXTEN_NOT_FOUND:
 			case SIP_GET_DEST_REFUSED:
 			default:
@@ -27244,7 +27284,12 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
 		res = 1;
 	} else if (!strcasecmp(v->name, "allowoverlap")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_ALLOWOVERLAP);
-		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWOVERLAP);
+		ast_clear_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP);
+		if (ast_true(v->value)) {
+			ast_set_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP_YES);
+		} else if (!strcasecmp(v->value, "dtmf")){
+			ast_set_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP_DTMF);
+		}
 	} else if (!strcasecmp(v->name, "allowsubscribe")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE);
@@ -28459,6 +28504,7 @@ static int reload_config(enum channelreloadreason reason)
 	sipdebug &= sip_debug_console;
 	ast_clear_flag(&global_flags[0], AST_FLAGS_ALL);
 	ast_clear_flag(&global_flags[1], AST_FLAGS_ALL);
+	ast_clear_flag(&global_flags[2], AST_FLAGS_ALL);
 
 	/* Reset IP addresses  */
 	ast_sockaddr_parse(&bindaddr, "0.0.0.0:0", 0);
@@ -28534,7 +28580,7 @@ static int reload_config(enum channelreloadreason reason)
 	sip_cfg.allowtransfer = TRANSFER_OPENFORALL;	/* Merrily accept all transfers by default */
 	sip_cfg.rtautoclear = 120;
 	ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE);	/* Default for all devices: TRUE */
-	ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP);		/* Default for all devices: TRUE */
+	ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP_YES);	/* Default for all devices: Yes */
 	sip_cfg.peer_rtupdate = TRUE;
 	global_dynamic_exclude_static = 0;	/* Exclude static peers */
 	sip_cfg.tcp_enabled = FALSE;
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index bc4ea4d96..0c3661d91 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -303,46 +303,52 @@
 	a second page of flags (for flags[1] */
 /*@{*/
 /* realtime flags */
-#define SIP_PAGE2_RTCACHEFRIENDS		(1 <<  0)    /*!< GP: Should we keep RT objects in memory for extended time? */
-#define SIP_PAGE2_RTAUTOCLEAR			(1 <<  1)    /*!< GP: Should we clean memory from peers after expiry? */
-#define SIP_PAGE2_RPID_UPDATE			(1 <<  2)
-#define SIP_PAGE2_Q850_REASON			(1 <<  3)    /*!< DP: Get/send cause code via Reason header */
-#define SIP_PAGE2_SYMMETRICRTP			(1 <<  4)    /*!< GDP: Whether symmetric RTP is enabled or not */
-#define SIP_PAGE2_STATECHANGEQUEUE		(1 <<  5)    /*!< D: Unsent state pending change exists */
-#define SIP_PAGE2_CONNECTLINEUPDATE_PEND	(1 <<  6)
-#define SIP_PAGE2_RPID_IMMEDIATE		(1 <<  7)
-#define SIP_PAGE2_RPORT_PRESENT			(1 <<  8)   /*!< Was rport received in the Via header? */
-#define SIP_PAGE2_PREFERRED_CODEC		(1 <<  9)   /*!< GDP: Only respond with single most preferred joint codec */
-#define SIP_PAGE2_VIDEOSUPPORT			(1 << 10)   /*!< DP: Video supported if offered? */
-#define SIP_PAGE2_TEXTSUPPORT			(1 << 11)   /*!< GDP: Global text enable */
-#define SIP_PAGE2_ALLOWSUBSCRIBE		(1 << 12)   /*!< GP: Allow subscriptions from this peer? */
-#define SIP_PAGE2_ALLOWOVERLAP			(1 << 13)   /*!< DP: Allow overlap dialing ? */
-#define SIP_PAGE2_SUBSCRIBEMWIONLY		(1 << 14)   /*!< GP: Only issue MWI notification if subscribed to */
-#define SIP_PAGE2_IGNORESDPVERSION		(1 << 15)   /*!< GDP: Ignore the SDP session version number we receive and treat all sessions as new */
-
-#define SIP_PAGE2_T38SUPPORT			(3 << 16)    /*!< GDP: T.38 Fax Support */
-#define SIP_PAGE2_T38SUPPORT_UDPTL		(1 << 16)    /*!< GDP: T.38 Fax Support (no error correction) */
-#define SIP_PAGE2_T38SUPPORT_UDPTL_FEC		(2 << 16)    /*!< GDP: T.38 Fax Support (FEC error correction) */
-#define SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY	(3 << 16)    /*!< GDP: T.38 Fax Support (redundancy error correction) */
-
-#define SIP_PAGE2_CALL_ONHOLD			(3 << 18)  /*!< D: Call hold states: */
-#define SIP_PAGE2_CALL_ONHOLD_ACTIVE		(1 << 18)  /*!< D: Active hold */
-#define SIP_PAGE2_CALL_ONHOLD_ONEDIR		(2 << 18)  /*!< D: One directional hold */
-#define SIP_PAGE2_CALL_ONHOLD_INACTIVE		(3 << 18)  /*!< D: Inactive hold */
-
-#define SIP_PAGE2_RFC2833_COMPENSATE		(1 << 20)  /*!< DP: Compensate for buggy RFC2833 implementations */
-#define SIP_PAGE2_BUGGY_MWI			(1 << 21)  /*!< DP: Buggy CISCO MWI fix */
-#define SIP_PAGE2_DIALOG_ESTABLISHED		(1 << 22)  /*!< 29: Has a dialog been established? */
-
-#define SIP_PAGE2_FAX_DETECT			(3 << 23)  /*!< DP: Fax Detection support */
-#define SIP_PAGE2_FAX_DETECT_CNG		(1 << 23)  /*!< DP: Fax Detection support - detect CNG in audio */
-#define SIP_PAGE2_FAX_DETECT_T38		(2 << 23)  /*!< DP: Fax Detection support - detect T.38 reinvite from peer */
-#define SIP_PAGE2_FAX_DETECT_BOTH		(3 << 23)  /*!< DP: Fax Detection support - detect both */
-
-#define SIP_PAGE2_UDPTL_DESTINATION		(1 << 25)  /*!< DP: Use source IP of RTP as destination if NAT is enabled */
-#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS		(1 << 26)  /*!< DP: Always set up video, even if endpoints don't support it */
-#define SIP_PAGE2_HAVEPEERCONTEXT	(1 << 27)	/*< Are we associated with a configured peer context? */
-#define SIP_PAGE2_USE_SRTP              (1 << 28)    /*!< DP: Whether we should offer (only)  SRTP */
+#define SIP_PAGE2_RTCACHEFRIENDS            (1 <<  0)   /*!< GP: Should we keep RT objects in memory for extended time? */
+#define SIP_PAGE2_RTAUTOCLEAR               (1 <<  1)   /*!< GP: Should we clean memory from peers after expiry? */
+#define SIP_PAGE2_RPID_UPDATE               (1 <<  2)
+#define SIP_PAGE2_Q850_REASON               (1 <<  3)   /*!< DP: Get/send cause code via Reason header */
+#define SIP_PAGE2_SYMMETRICRTP              (1 <<  4)   /*!< GDP: Whether symmetric RTP is enabled or not */
+#define SIP_PAGE2_STATECHANGEQUEUE          (1 <<  5)   /*!< D: Unsent state pending change exists */
+#define SIP_PAGE2_CONNECTLINEUPDATE_PEND    (1 <<  6)
+#define SIP_PAGE2_RPID_IMMEDIATE            (1 <<  7)
+#define SIP_PAGE2_RPORT_PRESENT             (1 <<  8)   /*!< Was rport received in the Via header? */
+#define SIP_PAGE2_PREFERRED_CODEC           (1 <<  9)   /*!< GDP: Only respond with single most preferred joint codec */
+#define SIP_PAGE2_VIDEOSUPPORT              (1 << 10)   /*!< DP: Video supported if offered? */
+#define SIP_PAGE2_TEXTSUPPORT               (1 << 11)   /*!< GDP: Global text enable */
+#define SIP_PAGE2_ALLOWSUBSCRIBE            (1 << 12)   /*!< GP: Allow subscriptions from this peer? */
+
+#define SIP_PAGE2_ALLOWOVERLAP              (3 << 13)   /*!< DP: Allow overlap dialing ? */
+#define SIP_PAGE2_ALLOWOVERLAP_NO           (0 << 13)   /*!< No, terminate with 404 Not found */
+#define SIP_PAGE2_ALLOWOVERLAP_YES          (1 << 13)   /*!< Yes, using the 484 Address Incomplete response */
+#define SIP_PAGE2_ALLOWOVERLAP_DTMF         (2 << 13)   /*!< Yes, using the DTMF transmission through Early Media */
+#define SIP_PAGE2_ALLOWOVERLAP_SPARE        (3 << 13)   /*!< Spare (reserved for another dialling transmission mechanisms like KPML) */
+
+#define SIP_PAGE2_SUBSCRIBEMWIONLY          (1 << 15)   /*!< GP: Only issue MWI notification if subscribed to */
+#define SIP_PAGE2_IGNORESDPVERSION          (1 << 16)   /*!< GDP: Ignore the SDP session version number we receive and treat all sessions as new */
+
+#define SIP_PAGE2_T38SUPPORT                (3 << 17)   /*!< GDP: T.38 Fax Support */
+#define SIP_PAGE2_T38SUPPORT_UDPTL          (1 << 17)   /*!< GDP: T.38 Fax Support (no error correction) */
+#define SIP_PAGE2_T38SUPPORT_UDPTL_FEC      (2 << 17)   /*!< GDP: T.38 Fax Support (FEC error correction) */
+#define SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY   (3 << 17)   /*!< GDP: T.38 Fax Support (redundancy error correction) */
+
+#define SIP_PAGE2_CALL_ONHOLD               (3 << 19)   /*!< D: Call hold states: */
+#define SIP_PAGE2_CALL_ONHOLD_ACTIVE        (1 << 19)   /*!< D: Active hold */
+#define SIP_PAGE2_CALL_ONHOLD_ONEDIR        (2 << 19)   /*!< D: One directional hold */
+#define SIP_PAGE2_CALL_ONHOLD_INACTIVE      (3 << 19)   /*!< D: Inactive hold */
+
+#define SIP_PAGE2_RFC2833_COMPENSATE        (1 << 21)   /*!< DP: Compensate for buggy RFC2833 implementations */
+#define SIP_PAGE2_BUGGY_MWI                 (1 << 22)   /*!< DP: Buggy CISCO MWI fix */
+#define SIP_PAGE2_DIALOG_ESTABLISHED        (1 << 23)   /*!< 29: Has a dialog been established? */
+
+#define SIP_PAGE2_FAX_DETECT                (3 << 24)   /*!< DP: Fax Detection support */
+#define SIP_PAGE2_FAX_DETECT_CNG            (1 << 24)   /*!< DP: Fax Detection support - detect CNG in audio */
+#define SIP_PAGE2_FAX_DETECT_T38            (2 << 24)   /*!< DP: Fax Detection support - detect T.38 reinvite from peer */
+#define SIP_PAGE2_FAX_DETECT_BOTH           (3 << 24)   /*!< DP: Fax Detection support - detect both */
+
+#define SIP_PAGE2_UDPTL_DESTINATION         (1 << 26)   /*!< DP: Use source IP of RTP as destination if NAT is enabled */
+#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS       (1 << 27)   /*!< DP: Always set up video, even if endpoints don't support it */
+#define SIP_PAGE2_HAVEPEERCONTEXT           (1 << 28)   /*< Are we associated with a configured peer context? */
+#define SIP_PAGE2_USE_SRTP                  (1 << 29)   /*!< DP: Whether we should offer (only)  SRTP */
 
 #define SIP_PAGE2_FLAGS_TO_COPY \
 	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
@@ -466,7 +472,7 @@ enum sip_auth_type {
 
 /*! \brief Result from get_destination function */
 enum sip_get_dest_result {
-	SIP_GET_DEST_PICKUP_EXTEN_FOUND = 1,
+	SIP_GET_DEST_EXTEN_MATCHMORE = 1,
 	SIP_GET_DEST_EXTEN_FOUND = 0,
 	SIP_GET_DEST_EXTEN_NOT_FOUND = -1,
 	SIP_GET_DEST_REFUSED = -2,
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 52800b17c..3c77a88be 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -122,6 +122,13 @@ context=default                 ; Default context for incoming calls
                                 ; 'username' field from the authentication line
                                 ; instead of the From: field.
 allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
+;allowoverlap=yes               ; Enable RFC3578 overlap dialing support.
+                                ; Can use the Incomplete application to collect the
+                                ; needed digits from an ambiguous dialplan match.
+;allowoverlap=dtmf              ; Enable overlap dialing support using DTMF delivery
+                                ; methods (inband, RFC2833, SIP INFO) in the early
+                                ; media phase.  Uses the Incomplete application to
+                                ; collect the needed digits.
 ;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
                                 ; Default is enabled. The Dial() options 't' and 'T' are not
                                 ; related as to whether SIP transfers are allowed or not.