[asterisk-commits] irroot: branch irroot/distrotech-customers-1.8 r345364 - in /team/irroot/dist...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Nov 15 08:25:06 CST 2011
Author: irroot
Date: Tue Nov 15 08:24:51 2011
New Revision: 345364
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=345364
Log:
Multiple revisions 345062-345063,345160,345163,345219,345273,345285
........
r345062 | jrose | 2011-11-14 17:00:05 +0200 (Mon, 14 Nov 2011) | 11 lines
Moves voicemail setup password entry to the end of the setup process.
This change was made because forcegreeting and forcename settings in voicemail could be
circumvented by hanging up after entering a password, because the only way voicemail
currently observes whether a mailbox is new or not is by checking to see if the password
is the same as the mailbox number or not.
(closes issue ASTERISK-18282)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1581/
........
r345063 | kmoore | 2011-11-14 17:08:12 +0200 (Mon, 14 Nov 2011) | 6 lines
Ensure that a null vmexten does not cause a segfault
When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten
was not expected to be null. This change handles that situation to avoid
a segfault.
........
r345160 | wdoekes | 2011-11-14 21:00:28 +0200 (Mon, 14 Nov 2011) | 9 lines
Update reqresp_parser parse_uri doxygen comments.
The issue mentioned in the bug report had been fixed recently by
twilson. The reporter included this documentation fix.
(closes issue ASTERISK-18572)
Reported by: Richard Miller
Patch by: Richard Miller (modified)
........
r345163 | twilson | 2011-11-14 21:05:09 +0200 (Mon, 14 Nov 2011) | 16 lines
Don't read past end of input when calling write()
int blah = 1;
...
write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) !=
(new_frames * sizeof(blah)))
is only valid when new_frames == 1. Otherwise we start reading into adjacent
variables declared on the stack. The read end discards what is read, so the
values don't matter but it's not a good idea to read past where we want even
though new_frames is almost always 1 and should never be large. This patch is
basically taken out of kpfleming's eventfd branch, as he mentioned that he
remembered fixing it there when I talked to him about this issue.
Review: https://reviewboard.asterisk.org/r/1583/
........
r345219 | rmudgett | 2011-11-14 22:45:49 +0200 (Mon, 14 Nov 2011) | 7 lines
Fix Progress spelling error in main/pbx.c.
(closes issue ASTERISK-18857)
Reported by: David M
Patches:
mainpbx-trivial.patch (License #6326) patch uploaded by David M
........
r345273 | rmudgett | 2011-11-14 23:43:39 +0200 (Mon, 14 Nov 2011) | 30 lines
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/
........
r345285 | rmudgett | 2011-11-15 00:19:18 +0200 (Tue, 15 Nov 2011) | 13 lines
Make queue log indicate if ADDMEMBER is paused for AMI and realtime.
* Add parameter to queue log ADDMEMBER to indicate if the member is
paused.
(closes issue ASTERISK-18645)
Reported by: garlew
Patches:
paused.diff (License #5337) patch uploaded by garlew
Tested by: rmudgett, garlew
Review: https://reviewboard.asterisk.org/r/1469/
........
Merged revisions 345062-345063,345160,345163,345219,345273,345285 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Modified:
team/irroot/distrotech-customers-1.8/ (props changed)
team/irroot/distrotech-customers-1.8/UPGRADE.txt
team/irroot/distrotech-customers-1.8/channels/chan_sip.c
team/irroot/distrotech-customers-1.8/channels/sip/include/sip.h
team/irroot/distrotech-customers-1.8/configs/sip.conf.sample
Propchange: team/irroot/distrotech-customers-1.8/
------------------------------------------------------------------------------
automerge = *
Propchange: team/irroot/distrotech-customers-1.8/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Nov 15 08:24:51 2011
@@ -1,1 +1,1 @@
-/branches/1.8:1-345230
+/branches/1.8:1-345362
Modified: team/irroot/distrotech-customers-1.8/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/team/irroot/distrotech-customers-1.8/UPGRADE.txt?view=diff&rev=345364&r1=345363&r2=345364
==============================================================================
--- team/irroot/distrotech-customers-1.8/UPGRADE.txt (original)
+++ team/irroot/distrotech-customers-1.8/UPGRADE.txt Tue Nov 15 08:24:51 2011
@@ -162,6 +162,10 @@
* 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
Modified: team/irroot/distrotech-customers-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/irroot/distrotech-customers-1.8/channels/chan_sip.c?view=diff&rev=345364&r1=345363&r2=345364
==============================================================================
--- team/irroot/distrotech-customers-1.8/channels/chan_sip.c (original)
+++ team/irroot/distrotech-customers-1.8/channels/chan_sip.c Tue Nov 15 08:24:51 2011
@@ -1367,6 +1367,7 @@
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);
@@ -6807,17 +6808,25 @@
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) &&
@@ -15189,18 +15198,23 @@
}
} 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
@@ -15219,11 +15233,12 @@
}
}
- /* 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;
@@ -16675,6 +16690,19 @@
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
@@ -17228,7 +17256,7 @@
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)" : "");
@@ -17782,7 +17810,7 @@
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)));
@@ -20961,10 +20989,13 @@
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;
@@ -21674,7 +21705,7 @@
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";
@@ -22395,12 +22426,21 @@
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:
@@ -26520,7 +26560,12 @@
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);
@@ -27706,6 +27751,7 @@
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);
@@ -27778,7 +27824,7 @@
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: team/irroot/distrotech-customers-1.8/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/team/irroot/distrotech-customers-1.8/channels/sip/include/sip.h?view=diff&rev=345364&r1=345363&r2=345364
==============================================================================
--- team/irroot/distrotech-customers-1.8/channels/sip/include/sip.h (original)
+++ team/irroot/distrotech-customers-1.8/channels/sip/include/sip.h Tue Nov 15 08:24:51 2011
@@ -300,46 +300,52 @@
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 | \
@@ -449,7 +455,7 @@
/*! \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: team/irroot/distrotech-customers-1.8/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/irroot/distrotech-customers-1.8/configs/sip.conf.sample?view=diff&rev=345364&r1=345363&r2=345364
==============================================================================
--- team/irroot/distrotech-customers-1.8/configs/sip.conf.sample (original)
+++ team/irroot/distrotech-customers-1.8/configs/sip.conf.sample Tue Nov 15 08:24:51 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.
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