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
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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
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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
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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
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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
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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
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276653
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010) | 1 line
Update documentation for voicemail.conf externpass option.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276268
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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
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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
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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
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r276126 | russell | 2010-07-13 14:14:54 -0500 (Tue, 13 Jul 2010) | 2 lines
Only reset a CDR that exists.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276127
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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
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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).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276124
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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
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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
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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
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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
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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
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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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276074
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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
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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
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275995
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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
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r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13 Jul 2010) | 2 lines
Move SQL scripts into their own database-specific directories.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275910
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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
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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
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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
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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
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275682
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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
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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
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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
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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
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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
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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
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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-
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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-
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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-
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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-
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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-
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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
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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
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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-
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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
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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
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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
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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
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275186
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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
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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
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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
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275144
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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
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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
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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
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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
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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
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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
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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
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<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
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<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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274595
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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
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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
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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
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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
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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
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274418
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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
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274316
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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
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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
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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
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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
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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
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274164
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Matthew Nicholson [Tue, 6 Jul 2010 13:52:56 +0000 (13:52 +0000)]
Blocked revisions 274093 via svnmerge
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274094
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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
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Tilghman Lesher [Mon, 5 Jul 2010 20:00:48 +0000 (20:00 +0000)]
Blocked revisions 273981 via svnmerge
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r273981 | tilghman | 2010-07-05 14:48:42 -0500 (Mon, 05 Jul 2010) | 2 lines
Command 'stop gracefully' doesn't.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273982
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Paul Belanger [Mon, 5 Jul 2010 13:53:44 +0000 (13:53 +0000)]
Merged revisions 273884 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul 2010) | 8 lines
Remove extra line breaks from 'core show config mappings'
(closes issue #17583)
Reported by: pabelanger
Patches:
issue17583.patch uploaded by pabelanger (license 224)
Tested by: lmadsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273886
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Tilghman Lesher [Sat, 3 Jul 2010 02:36:31 +0000 (02:36 +0000)]
Merged revisions 273793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407)
Reported by: pdf
Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273830
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Tilghman Lesher [Fri, 2 Jul 2010 17:10:59 +0000 (17:10 +0000)]
Merged revisions 273717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) | 8 lines
Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup.
(closes issue #17564)
Reported by: ramonpeek
Patches:
20100630__issue17564.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273718
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Tilghman Lesher [Fri, 2 Jul 2010 16:57:50 +0000 (16:57 +0000)]
Blocked revisions 273639 via svnmerge
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r273639 | tilghman | 2010-07-02 10:46:27 -0500 (Fri, 02 Jul 2010) | 8 lines
If all members are paused, the wrong status is indicated.
(closes issue #17576)
Reported by: ramonpeek
Patches:
diff.txt uploaded by ramonpeek (license 266)
Tested by: ramonpeek
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273715
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Tilghman Lesher [Fri, 2 Jul 2010 16:57:28 +0000 (16:57 +0000)]
The switch fallthrough could create some errorneous situations, so best to force directly to the default case.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273714
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Tzafrir Cohen [Fri, 2 Jul 2010 15:57:02 +0000 (15:57 +0000)]
Fix various typos reported by Lintian
(Also fix the typos in the comments)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273641
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Russell Bryant [Thu, 1 Jul 2010 22:16:23 +0000 (22:16 +0000)]
Merged revisions 273565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines
Don't return a partially initialized datastore.
If memory allocation fails in ast_strdup(), don't return a partially
initialized datastore. Bad things may happen.
(related to ABE-2415)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273566
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Jeff Peeler [Thu, 1 Jul 2010 20:28:15 +0000 (20:28 +0000)]
Merged revisions 273474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines
Allow admin user to join conference without using admin mode and no user pin.
Configuring the conference in meetme.conf like the following:
conf => 2345,,6666
did not prompt for pin when used without admin mode. This meant that the
conference could not be joined as an admin even if the user knew the correct
pin. The original bug report was submitted claiming that the blank user pin
should deny entry into the conference. I think a better way to handle this
would be with a feature enhancement that used the following syntax:
conf => 2345,X,6666 - where X denotes no acceptable pin allowed
(closes issue #15704)
Reported by: modelnine
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273522
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Matthew Nicholson [Thu, 1 Jul 2010 19:34:47 +0000 (19:34 +0000)]
Properly handle failures of fax->start_session()
FAX-177
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273464
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David Vossel [Thu, 1 Jul 2010 16:40:17 +0000 (16:40 +0000)]
correct handling of get_destination return values
A failure when calling the get_destination can mean multiple things. If
the extension is not found, a 404 error is appropriate, but if the URI
scheme is incorrect, a 404 is not approperiate. This patch adds the
get_destination_result enum to differentiate between these and other failure
types. The only logical difference in this patch is that we now send a "416
Unsupported URI scheme" response instead of a "404" when the scheme is not
recognized. This indicates to the initiator of the INVITE to retry the request
with a correct URI.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273427
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