asterisk/asterisk.git
9 years agoFormatting changes (guideline corrections)
Olle Johansson [Fri, 16 Jul 2010 10:31:42 +0000 (10:31 +0000)]
Formatting changes (guideline corrections)

Found a unused bag of curly brackets under my table. I always wondered where
they had gone. They where indeed needed in chan_sip.c

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276989 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdding a few more credits
Olle Johansson [Fri, 16 Jul 2010 10:08:45 +0000 (10:08 +0000)]
Adding a few more credits

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276952 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd ability to configure the Max-Forwards header in the dialplan, as well as in
Olle Johansson [Fri, 16 Jul 2010 10:00:58 +0000 (10:00 +0000)]
Add ability to configure the Max-Forwards header in the dialplan, as well as in
sip.conf configuration for the channel and for devices.

The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary,
like SIP proxys and SBCs, decrement this counter and detects when it reaches zero,
at which point the SIP request is nicely killed in a SIP-friendly way.

Review: https://reviewboard.asterisk.org/r/778/

Thanks to dvossel for the review and good advice.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd a dialplan function to check if a queue exists: QUEUE_EXISTS
Olle Johansson [Fri, 16 Jul 2010 09:25:48 +0000 (09:25 +0000)]
Add a dialplan function to check if a queue exists: QUEUE_EXISTS

Review: https://reviewboard.asterisk.org/r/777/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276950 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAnd yet one more
Tilghman Lesher [Fri, 16 Jul 2010 06:04:22 +0000 (06:04 +0000)]
And yet one more

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276911 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years ago"Item may be used uninitialized in this function."
Tilghman Lesher [Fri, 16 Jul 2010 05:59:11 +0000 (05:59 +0000)]
"Item may be used uninitialized in this function."

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix reversed logic of if statement.
Mark Michelson [Fri, 16 Jul 2010 05:42:24 +0000 (05:42 +0000)]
Fix reversed logic of if statement.

Found based on message from Philip Prindeville on the
Asterisk Developers mailing list.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276909 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDetect the --dynamic-list flag a bit better
Tilghman Lesher [Fri, 16 Jul 2010 05:38:06 +0000 (05:38 +0000)]
Detect the --dynamic-list flag a bit better

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276908 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix build on FreeBSD
Tilghman Lesher [Fri, 16 Jul 2010 04:45:33 +0000 (04:45 +0000)]
Fix build on FreeBSD

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276871 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix trunk build for Mac OS X 10.6
Tilghman Lesher [Fri, 16 Jul 2010 04:23:02 +0000 (04:23 +0000)]
Fix trunk build for Mac OS X 10.6

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAllow ipaddress to contain the maximum IPv6 address.
Tilghman Lesher [Fri, 16 Jul 2010 04:18:58 +0000 (04:18 +0000)]
Allow ipaddress to contain the maximum IPv6 address.

Also, update meetme to the full list of supported fields.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276869 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoQuote AC_SUBST within m4_ifval, so it does not get prematurely expanded.
Tilghman Lesher [Thu, 15 Jul 2010 23:25:09 +0000 (23:25 +0000)]
Quote AC_SUBST within m4_ifval, so it does not get prematurely expanded.

(closes issue #17654)
 Reported by: pprindeville
 Patches:
       issue17654.diff uploaded by qwell (license 4)
 Tested by: qwell, pprindeville

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276830 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoCorrect not setting the bindport before attempting to open the socket.
Jeff Peeler [Thu, 15 Jul 2010 20:21:03 +0000 (20:21 +0000)]
Correct not setting the bindport before attempting to open the socket.

Related to changes from 276571, I was accidentally testing with a port set in
my configuration causing me to miss this. Also moved the TCP handling as well
to occur before build_peer is called.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276788 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDefine LLONG_MAX on systems that do not have it.
Tilghman Lesher [Thu, 15 Jul 2010 19:46:57 +0000 (19:46 +0000)]
Define LLONG_MAX on systems that do not have it.

(closes issue #17644)
 Reported by: pprindeville

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276769 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix linking asterisk on CentOS 5, which is using gcc 4.1.1. Gcc 4.1.2 has the real...
Tilghman Lesher [Thu, 15 Jul 2010 18:44:20 +0000 (18:44 +0000)]
Fix linking asterisk on CentOS 5, which is using gcc 4.1.1.  Gcc 4.1.2 has the real fix.

Review: https://reviewboard.asterisk.org/r/790/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 276652 via svnmerge from
Jeff Peeler [Thu, 15 Jul 2010 13:51:11 +0000 (13:51 +0000)]
Merged revisions 276652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) | 2 lines

  In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276653 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd lua5.1 to the handy dandy list of packages.
Russell Bryant [Thu, 15 Jul 2010 12:21:10 +0000 (12:21 +0000)]
Add lua5.1 to the handy dandy list of packages.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276616 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix MWI notification transmission problems over SIP.
Jeff Peeler [Wed, 14 Jul 2010 22:58:24 +0000 (22:58 +0000)]
Fix MWI notification transmission problems over SIP.

MWI updates were not being sent if no messages were found in the event cache.
This was corrected since a phone may need to clear its MWI status configured
previously from another mailbox.

Upon module or sip reload, MWI updates could not be sent due to the sipsock
socket not being set early enough in reload_config. The code handling the
descriptor assignment and such has simply been moved before the call to
build_peer.

Issuing a sip reload cleared the IP address of the peer, but skipped checking
the database for registration information. The database is now checked both
for sip reload and actually reloading the module.

If a transmission occurs before the do_monitor thread has started, do not
attempt to send a signal to it.

(closes issue #17398)
Reported by: ip-rob

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276571 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix errors where incorrect address information was printed.
Mark Michelson [Wed, 14 Jul 2010 22:32:29 +0000 (22:32 +0000)]
Fix errors where incorrect address information was printed.

ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions)
uses thread-local storage for storing the string that it creates. In cases where
ast_sockaddr_stringify_fmt was being called twice within the same statement, the
result of one call would be overwritten by the result of the other call. This
usually was happening in printf-like statements and was resulting in the same
stringified addressed being printed twice instead of two separate addresses.

I have fixed this by using ast_strdupa on the result of stringify functions if
they are used twice within the same statement. As far as I could tell, there were
no instances where a pointer to the result of such a call were saved anywhere, so
this is the only situation I could see where this error could occur.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276570 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMake compile again.
Richard Mudgett [Wed, 14 Jul 2010 21:29:32 +0000 (21:29 +0000)]
Make compile again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276531 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoOops, merge reverted this fix.
Tilghman Lesher [Wed, 14 Jul 2010 21:11:09 +0000 (21:11 +0000)]
Oops, merge reverted this fix.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276493 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemove the old stub files, preferring the optional_api method.
Tilghman Lesher [Wed, 14 Jul 2010 20:48:59 +0000 (20:48 +0000)]
Remove the old stub files, preferring the optional_api method.

(closes issue #17475)
 Reported by: tilghman

Review: https://reviewboard.asterisk.org/r/695/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276490 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDon't try to call an embedded module's backup_globals() function until
Kevin P. Fleming [Wed, 14 Jul 2010 20:15:48 +0000 (20:15 +0000)]
Don't try to call an embedded module's backup_globals() function until
after confirming it exists.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276441 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agohandle special case were "200 Ok" to pending INVITE never receives ACK
David Vossel [Wed, 14 Jul 2010 19:51:08 +0000 (19:51 +0000)]
handle special case were "200 Ok" to pending INVITE never receives ACK

Unlike most responses, the 200 Ok to a pending INVITE Request is
acknowledged by an ACK Request.  If the ACK Request for this Response is not received
the previous behavior was to immediately destroy the dialog and hangup
the channel. Now in an effort to be more RFC compliant, instead of immediately
destroying the dialog during this special case, termination is done with a BYE Request
as the dialog is technically confirmed when the 200 Ok is sent even if the ACK is
never received.  The behavior of immediately hanging up the channel remains.
This only affects how dialog termination proceeds for this one special case.

RFC 3261 section 13.3.1.4
"If the server retransmits the 2xx response for 64*T1 seconds without receiving
an ACK, the dialog is confirmed, but the session SHOULD be terminated.  This is
accomplished with a BYE, as described in Section 15."

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276439 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoExpand the caller ANI field to an ast_party_id
Richard Mudgett [Wed, 14 Jul 2010 16:58:03 +0000 (16:58 +0000)]
Expand the caller ANI field to an ast_party_id

Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.

This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/

Review: https://reviewboard.asterisk.org/r/744/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agocollapse debug code in retrans_pkt into separate lines
David Vossel [Wed, 14 Jul 2010 16:40:42 +0000 (16:40 +0000)]
collapse debug code in retrans_pkt into separate lines

I've been working in this function a bunch lately, and
these huge debug strings are getting annoying.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276392 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMake compile again.
Richard Mudgett [Wed, 14 Jul 2010 16:39:18 +0000 (16:39 +0000)]
Make compile again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276391 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDo not skip sending MWI for a peer if an address is defined. Really just a merge...
Jeff Peeler [Wed, 14 Jul 2010 16:36:02 +0000 (16:36 +0000)]
Do not skip sending MWI for a peer if an address is defined. Really just a merge mistake from IPv6

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276389 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix documentation for pgsql cel and cdr, and slightly improve pgsql_cel.
Tim Ringenbach [Wed, 14 Jul 2010 16:09:11 +0000 (16:09 +0000)]
Fix documentation for pgsql cel and cdr, and slightly improve pgsql_cel.

Change the documented pgsql schema to use "timestamp" instead of "time",
as the latter is only a time without a date.

Added some missing columns for cel's pgsql schema, and corrected spelling
on some others. Updated cel's uniqueid size to be the same as the cdr.
Added id column to cel's pgsql schema and updated code to allow unknown
columns to get their default value instead of forcing 0 or empty string.

Added microseconds to the timestamp cel logs to pgsql.

Review: https://reviewboard.asterisk.org/r/734

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276349 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoast_callerid restructuring
Richard Mudgett [Wed, 14 Jul 2010 15:48:36 +0000 (15:48 +0000)]
ast_callerid restructuring

The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.

Eliminate struct ast_callerid and replace it with the following struct
organization:

struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};

The new organization adds some new information as well.

* The party name and number now have their own presentation value that can
be manipulated independently.  ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.

* The party name and number now have a valid flag.  Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.

* The party name now has a character set value.  SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.

* The dialed party now has a numbering plan value that could be useful to
have available.

The various channel drivers will need to be updated to support the new
core features as needed.  They have simply been converted to supply
current functionality at this time.

The following items of note were either corrected or enhanced:

* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.

* CALLERPRES() is now deprecated because the name and number have their
own presentation values.

* Fixed app_alarmreceiver.c write_metadata().  The workstring[] could
contain garbage.  It also can only contain the caller id number so using
ast_callerid_parse() on it is silly.  There was also a typo in the
CALLERNAME if test.

* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string.  ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string.  Then using
ast_shrink_phone_number() could alter it even more.

* Fixed caller ID name and number memory leak in chan_usbradio.c.

* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.

* Protected access to a caller channel with lock in chan_sip.c.

* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk().  Also made save all caller ID data instead of just the name
and number strings.

* Simplified cdr.c set_one_cid().  It hand coded the ast_callerid_merge()
function.

* Corrected some weirdness with app_privacy.c's use of caller
presentation.

Review: https://reviewboard.asterisk.org/r/702/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 276267 via svnmerge from
Leif Madsen [Wed, 14 Jul 2010 11:51:48 +0000 (11:51 +0000)]
Merged revisions 276267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010) | 1 line

  Update documentation for voicemail.conf externpass option.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276268 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agochan_sip: RFC compliant retransmission timeout
David Vossel [Tue, 13 Jul 2010 22:18:38 +0000 (22:18 +0000)]
chan_sip: RFC compliant retransmission timeout

Retransmission of packets should not be based on how many packets were
sent, but instead on a timeout period.  Depending on whether or not the
packet is for a INVITE or NON-INVITE transaction, the number of packets
sent during the retransmission timeout period will be different, so
timing out based on the number of packets sent is not accurate.

This patch fixes this by removing the retransmit limit and only stopping
retransmission after a timeout period is reached.  By default this
timeout period is 64*(Timer T1) for both INVITE and non-INVITE
transactions.  For more information on sip timer values refer to
RFC3261 Appendix A.

Review: https://reviewboard.asterisk.org/r/749/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRevert early destruction of RTP sessions
Terry Wilson [Tue, 13 Jul 2010 21:42:42 +0000 (21:42 +0000)]
Revert early destruction of RTP sessions

Some code improperly assumes that the sessions are still there, so revert the
change until I can find all of them and fix them.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276206 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRecorded merge of revisions 276126 via svnmerge from
Russell Bryant [Tue, 13 Jul 2010 19:15:47 +0000 (19:15 +0000)]
Recorded merge of revisions 276126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r276126 | russell | 2010-07-13 14:14:54 -0500 (Tue, 13 Jul 2010) | 2 lines

  Only reset a CDR that exists.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276127 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 276123 via svnmerge from
Russell Bryant [Tue, 13 Jul 2010 19:09:42 +0000 (19:09 +0000)]
Merged revisions 276123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010) | 2 lines

  Use chan->cdr instead of chan_cdr (just like peer->cdr instead of peer_cdr in the last commit).
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276124 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoOops, XML documentation fix.
Tilghman Lesher [Tue, 13 Jul 2010 19:05:17 +0000 (19:05 +0000)]
Oops, XML documentation fix.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276122 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoIt really cannot fail in the places below, but the stupid compiler doesn't know that.
Tilghman Lesher [Tue, 13 Jul 2010 19:00:02 +0000 (19:00 +0000)]
It really cannot fail in the places below, but the stupid compiler doesn't know that.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276120 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoWeird compiler error on Bamboo.
Tilghman Lesher [Tue, 13 Jul 2010 18:41:59 +0000 (18:41 +0000)]
Weird compiler error on Bamboo.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276118 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFILE() now supports line-mode and writing (altering) files.
Tilghman Lesher [Tue, 13 Jul 2010 18:31:41 +0000 (18:31 +0000)]
FILE() now supports line-mode and writing (altering) files.

(closes issue #16461)
 Reported by: skyman
 Patches:
       20100622__issue16461.diff.txt uploaded by tilghman (license 14)
 Tested by: tilghman

Review: https://reviewboard.asterisk.org/r/737/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276114 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 275773 via svnmerge from
Jeff Peeler [Tue, 13 Jul 2010 17:37:40 +0000 (17:37 +0000)]
Merged revisions 275773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines

  Make user removals and traversals thread safe in meetme.

  Race conditions present in meetme involving the user list where a lack of
  locking has the potential for a user to be removed during a traversal or as in
  the case of the reporter after checking if the list is empty could cause a
  crash. Fixing this was done by convering the userlist to an ao2 container.

  (closes issue #17390)
  Reported by: Vince

  Review: https://reviewboard.asterisk.org/r/746/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276074 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDestroy RTP fds when we schedule final dialog destruction
Terry Wilson [Tue, 13 Jul 2010 17:11:37 +0000 (17:11 +0000)]
Destroy RTP fds when we schedule final dialog destruction

Since we are only keeping the dialog around for retransmissions at this point
and there is no possibility that we are still handling RTP, go ahead and
destroy the RTP sessions. Keeping them alive for 32 past when they are used
is unnecessary and can lead to problems with having too many open file
descriptors, etc.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275998 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 275994 via svnmerge from
Russell Bryant [Tue, 13 Jul 2010 16:53:44 +0000 (16:53 +0000)]
Merged revisions 275994 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010) | 14 lines

  Access peer->cdr directly instead of through a saved off reference.

  At this point in the code, it is possible that peer_cdr may be invalid.
  Specifically, in the blind transfer code, CDRs are swapped between channels.
  So, peer_cdr is no longer == peer->cdr.

  The scenario that exposed a crash in this code was a blind transfer that hit
  the system call limit, causing the transferee channel to get destroyed after
  the transfer attempt failed.  Even if it succeeds and this code doesn't crash,
  this code was still trying to reset a CDR on a channel that was now owned by
  a different thread, which is a BadThing(tm).

  (ABE-2417)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275995 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 275909 via svnmerge from
Tilghman Lesher [Tue, 13 Jul 2010 14:48:40 +0000 (14:48 +0000)]
Merged revisions 275909 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13 Jul 2010) | 2 lines

  Move SQL scripts into their own database-specific directories.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd example script for use with the externpasscheck voicemail.conf option.
Russell Bryant [Tue, 13 Jul 2010 11:41:54 +0000 (11:41 +0000)]
Add example script for use with the externpasscheck voicemail.conf option.

(closes issue #17628)
Reported by: lmadsen
Tested by: russell, lmadsen

Review: https://reviewboard.asterisk.org/r/774/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275863 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDon't try to ref authpeer when it isn't set
Terry Wilson [Mon, 12 Jul 2010 23:27:42 +0000 (23:27 +0000)]
Don't try to ref authpeer when it isn't set

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275816 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd which ITU spec specifies the numbering plan.
Richard Mudgett [Mon, 12 Jul 2010 17:54:46 +0000 (17:54 +0000)]
Add which ITU spec specifies the numbering plan.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275725 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 275665 via svnmerge from
Jeff Peeler [Mon, 12 Jul 2010 17:21:01 +0000 (17:21 +0000)]
Merged revisions 275665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010) | 11 lines

  Change ast_write to not stop generator when called from ast_prod.

  For SIP channels configured with the progressinband option on, the ringback was
  being immediately stopped. This problem was due to ast_prod being moved for a
  deadlock fix in 259858. Prodding the channel after setting up the generator
  triggered the check in ast_write to stop the generator. The fix here should
  write the frame the same as was done before the call to ast_prod was moved.

  (closes issue #17372)
  Reported by: tech_admin
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275682 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agocdr_pgsql does not detect when a table is found.
Leif Madsen [Mon, 12 Jul 2010 15:37:01 +0000 (15:37 +0000)]
cdr_pgsql does not detect when a table is found.

This change adds an ERROR message to let you know when a failure exists to
get the columns from the pgsql database, which typically means that the
table does not exist.

(closes issue #17478)
Reported by: kobaz
Patches:
      cdr_pgsql.patch uploaded by kobaz (license 834)
Tested by: kobaz, russell, lmadsen

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275626 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAllow netsock2.c to compile on systems that do not define AI_NUMERICSERV.
Mark Michelson [Mon, 12 Jul 2010 14:55:23 +0000 (14:55 +0000)]
Allow netsock2.c to compile on systems that do not define AI_NUMERICSERV.

(closes issue #17617)
Reported by: pprindeville
Patches:
      asterisk-trunk-bugid17617.patch uploaded by pprindeville (license 347)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275587 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdded support for indirect work mode.
TransNexus OSP Development [Mon, 12 Jul 2010 04:16:18 +0000 (04:16 +0000)]
Added support for indirect work mode.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275551 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoWhen creating a conference for a unit test, it is not mandatory to open a
Eliel C. Sardanons [Sat, 10 Jul 2010 20:49:30 +0000 (20:49 +0000)]
When creating a conference for a unit test, it is not mandatory to open a
dahdi pseudo channel, so if we fail doing it, continue creating the conference.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275509 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMake indentation consistent, move some queue features to the queue section.
Russell Bryant [Sat, 10 Jul 2010 14:48:03 +0000 (14:48 +0000)]
Make indentation consistent, move some queue features to the queue section.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275467 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd support for devices with less than 3 lines on the LCD.
Russell Bryant [Sat, 10 Jul 2010 14:44:18 +0000 (14:44 +0000)]
Add support for devices with less than 3 lines on the LCD.

(closes issue #17600)
Reported by: minaguib
Patches:
      ast_unistim_height_v2.patch uploaded by minaguib (license 1078)
Tested by: minaguib

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275466 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix some issues related to dynamic feature groups in features.conf.
Russell Bryant [Fri, 9 Jul 2010 21:57:21 +0000 (21:57 +0000)]
Fix some issues related to dynamic feature groups in features.conf.

The bridge handling code did not properly consider feature groups when setting
parameters that would affect whether or not a native bridge would be attempted.
If DYNAMIC_FEATURES only include a feature group, a native bridge would occur
that may prevent features from working.

Fix a bug in verbose output that would show the key mapping as empty if it was
using the default mapping and not a custom mapping in the feature group.

Add feature groups to the output of "features show".

Adjust the feature execution logic to match that of the logic when executing
a feature that was not configured through a feature group.

Update features.conf.sample to show that an '=' is still required if using
the default key mapping from [applicationmap].

Finally, clean up a little bit of formatting to better coform to coding
guidelines while in the area.

(closes issue #17589)
Reported by: lmadsen
Patches:
      issue_17589.rev4.txt uploaded by russell (license 2)
Tested by: russell, lmadsen

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275424 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix error in parsing SIP registry strings from ASTdb.
Mark Michelson [Fri, 9 Jul 2010 20:58:52 +0000 (20:58 +0000)]
Fix error in parsing SIP registry strings from ASTdb.

It was essentially an off-by-one error. The easiest way
to fix this was to use the handy-dandy AST_NONSTANDARD_RAW_ARGS
macro to parse the pieces of the registration string out. Tested
and it works wonderfully.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275385 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoGet more information about the Bamboo test failures
Tilghman Lesher [Fri, 9 Jul 2010 20:01:01 +0000 (20:01 +0000)]
Get more information about the Bamboo test failures

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275312 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd missing ao2_iterator_destroy().
Russell Bryant [Fri, 9 Jul 2010 19:58:06 +0000 (19:58 +0000)]
Add missing ao2_iterator_destroy().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275310 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix compile error.
Russell Bryant [Fri, 9 Jul 2010 19:56:41 +0000 (19:56 +0000)]
Fix compile error.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275309 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix port parsing in check_via.
Mark Michelson [Fri, 9 Jul 2010 19:46:20 +0000 (19:46 +0000)]
Fix port parsing in check_via.

If a Via header contained an IPv6 address, we would not properly parse
the port. We would instead get the information after the first colon in
the address.

(closes issue #17614)
Reported by: oej
Patches:
      diff uploaded by sperreault (license 252)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275308 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoInclude rdnis in msgXXXX.txt file.
Paul Belanger [Fri, 9 Jul 2010 19:32:47 +0000 (19:32 +0000)]
Include rdnis in msgXXXX.txt file.

(closes issue #17566)
Reported by: outcast
Patches:
      voicemail-rdnis.patch uploaded by outcast (license 1071)
Tested by: outcast

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275307 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix an issue where the port for p->ourip was being set to 0.
Mark Michelson [Fri, 9 Jul 2010 19:29:30 +0000 (19:29 +0000)]
Fix an issue where the port for p->ourip was being set to 0.

This should fix all the CDR tests that were not passing. When they would
originate a call, all fields in the INVITE that contained the source port would
have the port set to 0. Most troubling of these was the Contact header. Tests
are passing locally now and should also pass on the bamboo build agents.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275294 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 275241 via svnmerge from
Paul Belanger [Fri, 9 Jul 2010 19:21:27 +0000 (19:21 +0000)]
Merged revisions 275241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul 2010) | 8 lines

  Fix logging message for stale nonce.

  (closes issue #17582)
  Reported by: kenner
  Patches:
        chan_sip.c.diff uploaded by kenner (license 1040)
  Tested by: lmadsen
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275249 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoWeird, no output and Bamboo still fails...
Tilghman Lesher [Fri, 9 Jul 2010 18:55:02 +0000 (18:55 +0000)]
Weird, no output and Bamboo still fails...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 275182 via svnmerge from
Matthew Nicholson [Fri, 9 Jul 2010 18:24:03 +0000 (18:24 +0000)]
Merged revisions 275182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul 2010) | 2 lines

  give a better error message when attempting to unload a module that is not loaded
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275186 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd some diagnostic feedback to our data tests
Tilghman Lesher [Fri, 9 Jul 2010 18:21:39 +0000 (18:21 +0000)]
Add some diagnostic feedback to our data tests

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275172 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMove parking lot sample config out from the middle of dynamic features sample config.
Russell Bryant [Fri, 9 Jul 2010 18:11:13 +0000 (18:11 +0000)]
Move parking lot sample config out from the middle of dynamic features sample config.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275147 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 275143 via svnmerge from
Matthew Nicholson [Fri, 9 Jul 2010 17:50:45 +0000 (17:50 +0000)]
Merged revisions 275143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul 2010) | 2 lines

  don't unload modules that returned AST_MODULE_LOAD_DECLINE when they were loaded
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275144 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoKill some startup warnings and errors and make some messages more helpful in tracking...
Tilghman Lesher [Fri, 9 Jul 2010 17:00:22 +0000 (17:00 +0000)]
Kill some startup warnings and errors and make some messages more helpful in tracking down the source.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275105 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoReturn logic of sip_debug_test_addr() to its original functionality.
Mark Michelson [Fri, 9 Jul 2010 16:39:16 +0000 (16:39 +0000)]
Return logic of sip_debug_test_addr() to its original functionality.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275104 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 275027 via svnmerge from
Matthew Nicholson [Fri, 9 Jul 2010 16:05:58 +0000 (16:05 +0000)]
Merged revisions 275027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul 2010) | 8 lines

  Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial

  (closes issue #17592)
  Reported by: jamicque
  Patches:
        G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
  Tested by: jamicque, mnicholson
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275028 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 275021 via svnmerge from
Russell Bryant [Fri, 9 Jul 2010 15:35:53 +0000 (15:35 +0000)]
Merged revisions 275021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) | 4 lines

  Document that a leading and trailing slash is expected for test categories.

  Also, emit a warning if a test is registered without one of these.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275022 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix sip_uri_parse test comparison.
Mark Michelson [Fri, 9 Jul 2010 14:27:07 +0000 (14:27 +0000)]
Fix sip_uri_parse test comparison.

Part of the change with the IPv6 changes is to treat a host:port as
a single 'domain' entity. This test was not updated to have the correct
expectation after calling parse_uri().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274984 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoCopy the address into the peer structure after we set the default port
<simon.perreault@viagenie.ca> [Fri, 9 Jul 2010 13:30:37 +0000 (13:30 +0000)]
Copy the address into the peer structure after we set the default port

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSadly we can't dereference a pointer cast and use it as an lvalue without getting...
<simon.perreault@viagenie.ca> [Fri, 9 Jul 2010 12:56:18 +0000 (12:56 +0000)]
Sadly we can't dereference a pointer cast and use it as an lvalue without getting this
warning (at least with gcc 4.4.4):

netsock2.c:492: warning: dereferencing pointer ‘({anonymous})’ does break strict-aliasing rules

So we're back to using memcpy()...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274909 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoExtend length limit on country name in indications.conf.
Russell Bryant [Fri, 9 Jul 2010 12:48:25 +0000 (12:48 +0000)]
Extend length limit on country name in indications.conf.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274907 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMake it possible to disable individual cdr files per accountcode in cdr_csv
Olle Johansson [Fri, 9 Jul 2010 11:06:19 +0000 (11:06 +0000)]
Make it possible to disable individual cdr files per accountcode in cdr_csv

Review: https://reviewboard.asterisk.org/r/678/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274866 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix calls of ast_sockaddr_from_sin() from IPv6 integration.
Richard Mudgett [Thu, 8 Jul 2010 23:46:20 +0000 (23:46 +0000)]
Fix calls of ast_sockaddr_from_sin() from IPv6 integration.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274828 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix compile of chan_ooh323.c from IPv6 integration.
Richard Mudgett [Thu, 8 Jul 2010 23:23:17 +0000 (23:23 +0000)]
Fix compile of chan_ooh323.c from IPv6 integration.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274827 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAnd the automerge property.
Mark Michelson [Thu, 8 Jul 2010 22:16:16 +0000 (22:16 +0000)]
And the automerge property.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274786 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDelete properties I merged during v6-new merge.
Mark Michelson [Thu, 8 Jul 2010 22:15:25 +0000 (22:15 +0000)]
Delete properties I merged during v6-new merge.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274785 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd IPv6 to Asterisk.
Mark Michelson [Thu, 8 Jul 2010 22:08:07 +0000 (22:08 +0000)]
Add IPv6 to Asterisk.

This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.

Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.

(closes issue #17565)
Reported by: russell
Patches:
      asteriskv6-test-report.pdf uploaded by russell (license 2)

Review: https://reviewboard.asterisk.org/r/743

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoGenerate a correct AstData string for ast_callerid.cid_ton
Richard Mudgett [Thu, 8 Jul 2010 22:05:40 +0000 (22:05 +0000)]
Generate a correct AstData string for ast_callerid.cid_ton

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274782 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix trunk compile.
Richard Mudgett [Thu, 8 Jul 2010 19:12:55 +0000 (19:12 +0000)]
Fix trunk compile.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274773 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoImplement AstData API data providers as part of the GSOC 2010 project,
Eliel C. Sardanons [Thu, 8 Jul 2010 14:48:42 +0000 (14:48 +0000)]
Implement AstData API data providers as part of the GSOC 2010 project,
midterm evaluation.

Review: https://reviewboard.asterisk.org/r/757/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274727 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFixes some ref count issues introduced by r274539
David Vossel [Wed, 7 Jul 2010 20:09:00 +0000 (20:09 +0000)]
Fixes some ref count issues introduced by r274539

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274686 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd missing conditional around chan_dahdi mfcr2_skip_category config parameter.
Richard Mudgett [Wed, 7 Jul 2010 18:32:35 +0000 (18:32 +0000)]
Add missing conditional around chan_dahdi mfcr2_skip_category config parameter.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274639 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 274579 via svnmerge from
Richard Mudgett [Wed, 7 Jul 2010 18:20:00 +0000 (18:20 +0000)]
Merged revisions 274579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 Jul 2010) | 1 line

  Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274595 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSet proper FAXOPT(status), FAXOPT(statusstr), and FAXOPT(error) values where possible...
Matthew Nicholson [Wed, 7 Jul 2010 16:40:19 +0000 (16:40 +0000)]
Set proper FAXOPT(status), FAXOPT(statusstr), and FAXOPT(error) values where possible.  Previously some failure cases did not result in proper FAXOPT values.

FAX-203

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274540 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUse the relatedpeer field of a sip_pvt during INVITE processing.
Mark Michelson [Wed, 7 Jul 2010 16:21:53 +0000 (16:21 +0000)]
Use the relatedpeer field of a sip_pvt during INVITE processing.

Review: https://reviewboard.asterisk.org/r/629

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274539 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoChanged OSP TCP port from 1080 to 5045.
TransNexus OSP Development [Wed, 7 Jul 2010 07:07:08 +0000 (07:07 +0000)]
Changed OSP TCP port from 1080 to 5045.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274492 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAlso run the externnotify script when the pollmailboxes thread notices a change.
Tilghman Lesher [Wed, 7 Jul 2010 06:32:39 +0000 (06:32 +0000)]
Also run the externnotify script when the pollmailboxes thread notices a change.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274491 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 274417 via svnmerge from
Tilghman Lesher [Wed, 7 Jul 2010 06:15:43 +0000 (06:15 +0000)]
Merged revisions 274417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010) | 8 lines

  Correct how 100, 200, 300, etc. is said.  Also add the crazy British numbers.

  (closes issue #16102)
   Reported by: Delvar
   Patches:
         say.conf.fix.patch uploaded by Delvar (license 908)
         (plus a few additional fixes and simplifications by me)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274418 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 274283 via svnmerge from
Jeff Peeler [Tue, 6 Jul 2010 22:23:35 +0000 (22:23 +0000)]
Merged revisions 274283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines

  Correct sip.conf.sample comments for prematuremedia option.

  (closes issue #17513)
  Reported by: festr
  Patches:
        patch uploaded by festr (license 443)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 274280 via svnmerge from
Terry Wilson [Tue, 6 Jul 2010 22:15:27 +0000 (22:15 +0000)]
Merged revisions 274280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines

  Add option to not do a call forward on 482 Loop Detected

  Asterisk has always set up a forwarded call when receiving a 482 Loop Detected.
  This prevents handling the call failure by just continuing on in the dialplan.
  Since this would be a change in behavior, the new option to disable this
  behavior is forwardloopdetected which defaults to 'yes'.

  Review: https://reviewboard.asterisk.org/r/764/
........

(no option for trunk, just changing the behavior)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoStatus shows all non-CRC4 lines as "yellow", even if "yellow" was not in the bitfield.
Tilghman Lesher [Tue, 6 Jul 2010 22:09:23 +0000 (22:09 +0000)]
Status shows all non-CRC4 lines as "yellow", even if "yellow" was not in the bitfield.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274281 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoProperly detect and report invalid maxrate and maxrate values in the FAXOPT dialplan...
Matthew Nicholson [Tue, 6 Jul 2010 19:53:04 +0000 (19:53 +0000)]
Properly detect and report invalid maxrate and maxrate values in the FAXOPT dialplan function.  Also make fax_rate_str_to_int() return an unsigned int and return 0 instead of -1 in the event of an error.

FAX-202

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274243 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 274157 via svnmerge from
Mark Michelson [Tue, 6 Jul 2010 14:31:13 +0000 (14:31 +0000)]
Merged revisions 274157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul 2010) | 16 lines

  Fix problem with RFC 2833 DTMF not being accepted.

  A recent check was added to ensure that we did not erroneously
  detect duplicate DTMF when we received packets out of order.
  The problem was that the check did not account for the fact that
  the seqno of an RTP stream will roll over back to 0 after hitting
  65535. Now, we have a secondary check that will ensure that the
  seqno rolling over will not cause us to stop accepting DTMF.

  (closes issue #17571)
  Reported by: mdeneen
  Patches:
        rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
  Tested by: richardf, maxochoa, JJCinAZ
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274164 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoBlocked revisions 274093 via svnmerge
Matthew Nicholson [Tue, 6 Jul 2010 13:52:56 +0000 (13:52 +0000)]
Blocked revisions 274093 via svnmerge

........
  r274093 | mnicholson | 2010-07-06 08:52:28 -0500 (Tue, 06 Jul 2010) | 2 lines

  Make get_member_status return QUEUE_NO_MEMBERS instead of QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work again.  This regression was introduced in 273639.  Also fixed whitespace.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274094 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUh, yeah.
Tilghman Lesher [Tue, 6 Jul 2010 06:01:37 +0000 (06:01 +0000)]
Uh, yeah.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274053 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoBlocked revisions 273981 via svnmerge
Tilghman Lesher [Mon, 5 Jul 2010 20:00:48 +0000 (20:00 +0000)]
Blocked revisions 273981 via svnmerge

........
  r273981 | tilghman | 2010-07-05 14:48:42 -0500 (Mon, 05 Jul 2010) | 2 lines

  Command 'stop gracefully' doesn't.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273982 65c4cc65-6c06-0410-ace0-fbb531ad65f3