[svn-commits] rmudgett: branch 10 r345275 - in /branches/10: ./ channels/ channels/sip/incl...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Nov 14 15:59:35 CST 2011
Author: rmudgett
Date: Mon Nov 14 15:59:28 2011
New Revision: 345275
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=345275
Log:
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
Modified:
branches/10/ (props changed)
branches/10/UPGRADE-1.8.txt
branches/10/channels/chan_sip.c
branches/10/channels/sip/include/sip.h
branches/10/configs/sip.conf.sample
Propchange: branches/10/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.
Modified: branches/10/UPGRADE-1.8.txt
URL: http://svnview.digium.com/svn/asterisk/branches/10/UPGRADE-1.8.txt?view=diff&rev=345275&r1=345274&r2=345275
==============================================================================
--- branches/10/UPGRADE-1.8.txt (original)
+++ branches/10/UPGRADE-1.8.txt Mon Nov 14 15:59:28 2011
@@ -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 @@
* 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
@@ -154,6 +158,13 @@
If your outgoing folder is on a NFS mount or another network file system,
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:
@@ -249,6 +260,11 @@
* 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:
@@ -311,6 +327,3 @@
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.
-
Modified: branches/10/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/10/channels/chan_sip.c?view=diff&rev=345275&r1=345274&r2=345275
==============================================================================
--- branches/10/channels/chan_sip.c (original)
+++ branches/10/channels/chan_sip.c Mon Nov 14 15:59:28 2011
@@ -1381,6 +1381,7 @@
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);
@@ -6834,17 +6835,25 @@
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;
- }
- res = 0;
+ p->invitestate = INV_COMPLETED;
+ sip_alreadygone(p);
+ ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
+ break;
+ }
+ }
break;
case AST_CONTROL_PROCEEDING:
if ((ast->_state != AST_STATE_UP) &&
@@ -15487,18 +15496,23 @@
}
} 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
@@ -15517,11 +15531,12 @@
}
}
- /* 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;
@@ -17128,6 +17143,19 @@
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
@@ -17675,7 +17703,7 @@
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)" : "");
@@ -18230,7 +18258,7 @@
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)));
@@ -21497,10 +21525,13 @@
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;
@@ -22213,7 +22244,7 @@
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";
@@ -22934,12 +22965,21 @@
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:
@@ -27182,7 +27222,12 @@
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);
@@ -28388,6 +28433,7 @@
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);
@@ -28463,7 +28509,7 @@
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;
Modified: branches/10/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/branches/10/channels/sip/include/sip.h?view=diff&rev=345275&r1=345274&r2=345275
==============================================================================
--- branches/10/channels/sip/include/sip.h (original)
+++ branches/10/channels/sip/include/sip.h Mon Nov 14 15:59:28 2011
@@ -302,46 +302,52 @@
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 | \
@@ -465,7 +471,7 @@
/*! \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,
Modified: branches/10/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/10/configs/sip.conf.sample?view=diff&rev=345275&r1=345274&r2=345275
==============================================================================
--- branches/10/configs/sip.conf.sample (original)
+++ branches/10/configs/sip.conf.sample Mon Nov 14 15:59:28 2011
@@ -122,6 +122,13 @@
; '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.
More information about the svn-commits
mailing list