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r316336 | russell | 2011-05-03 17:13:31 -0500 (Tue, 03 May 2011) | 8 lines
Use htons() instead of ntohs() in some places.
(closes issue #19200)
Reported by: wdoekes
Patches:
issue19200-trunk.patch uploaded by wdoekes (license 717)
issue19200-1.8.x.patch uploaded by wdoekes (license 717)
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r316334 | dvossel | 2011-05-03 17:05:59 -0500 (Tue, 03 May 2011) | 8 lines
Fixes framehook segfault on indicate
(closes issue #19215)
Reported by: irroot
Patches:
framehook_indicate.patch uploaded by irroot (license 52)
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r316331 | russell | 2011-05-03 16:41:11 -0500 (Tue, 03 May 2011) | 2 lines
Resolve another warning.
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r316330 | dvossel | 2011-05-03 16:37:59 -0500 (Tue, 03 May 2011) | 24 lines
Merged revisions 316329 via svnmerge from
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r316329 | dvossel | 2011-05-03 16:29:55 -0500 (Tue, 03 May 2011) | 17 lines
Merged revisions 316328 via svnmerge from
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r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011) | 10 lines
Fixes chan_local crashs in local_fixup()
Thanks OEJ for tracking down the issue and submitting the patch.
(closes issue #19053)
Reported by: oej
Tested by: oej
Review: https://reviewboard.asterisk.org/r/1158/
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r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines
Fix a bunch of compiler warnings generated by gcc 4.6.0.
Most of these are -Wunused-but-set-variable, but there were a few others
mixed in here, as well.
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r316224 | rmudgett | 2011-05-03 14:18:30 -0500 (Tue, 03 May 2011) | 16 lines
The dahdi_hangup() call does not clean up the channel fully.
After dahdi_hangup() has supposedly hungup an ISDN channel there is still
traffic on the S0-bus because the channel was not cleaned up fully.
Shuffled the hangup code to include some missing cleanup. Also fixed some
code formatting in the area. I think the primary missing clean up code
was the call to tone_zone_play_tone() to turn off any active tones on the
channel.
(closes issue #19188)
Reported by: jg1234
Patches:
issue19188_v1.8.patch uploaded by rmudgett (license 664)
Tested by: jg1234
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r316217 | dvossel | 2011-05-03 13:59:06 -0500 (Tue, 03 May 2011) | 9 lines
Never put the Require: timer header in an Invite.
This has already been discussed and should have been resolved earlier. View
revsion 285565's log for more information about why it is important to not
put timer in the Require header.
(closes issue #18704)
Reported by: mfrager
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r316215 | dvossel | 2011-05-03 13:49:48 -0500 (Tue, 03 May 2011) | 9 lines
Fixes a random crash (NULL reference) in res_odbc.c.
(closes issue #19180)
Reported by: pruiz
Patches:
tmp.diff uploaded by pruiz (license 1152)
Tested by: pruiz, seanbright
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r316206 | seanbright | 2011-05-03 14:17:36 -0400 (Tue, 03 May 2011) | 8 lines
If we aren't interested in events, don't generate the FullyBooted event on AMI login.
(closes issue #19089)
Reported by: bklang
Patches:
issue19089-1.8-r316204.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r316094 | tilghman | 2011-05-02 14:09:55 -0500 (Mon, 02 May 2011) | 15 lines
Merged revisions 316093 via svnmerge from
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r316093 | tilghman | 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011) | 8 lines
More possible crashes based upon invalid inputs.
(closes issue #18161)
Reported by: wdoekes
Patches:
20110301__issue18161.diff.txt uploaded by tilghman (license 14)
Tested by: wdoekes
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r315894 | mnicholson | 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) | 28 lines
Merged revisions 315893 via svnmerge from
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r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines
Merged revisions 315891 via svnmerge from
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r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines
Fix our compliance with RFC 3261 section 18.2.2.
This change optimizes the free_via() function and removes some redundant null
checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
the port specified in the Via header for routing responses (even when maddr is
not set). Also the htons() function is now used when setting the port.
Additional documentation comments have been added in various places to make the
logic in the code clearer.
(closes issue #18951)
Reported by: jmls
Patches:
issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified)
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than beep and beeperr.
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r315810 | russell | 2011-04-27 10:55:48 -0500 (Wed, 27 Apr 2011) | 2 lines
Set the copyright year to 2011 in the startup message.
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r315765 | lmadsen | 2011-04-27 07:36:17 -0500 (Wed, 27 Apr 2011) | 4 lines
Enable Russian core sound selection in menuselect.
(closes issue #18724)
Reported by: pbxware
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r315673 | twilson | 2011-04-26 15:56:19 -0700 (Tue, 26 Apr 2011) | 25 lines
Merged revisions 315672 via svnmerge from
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r315672 | twilson | 2011-04-26 15:52:25 -0700 (Tue, 26 Apr 2011) | 18 lines
Merged revisions 315671 via svnmerge from
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r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) | 11 lines
Make sure unregistering a peer unlinks it from the peer container
Instead of mostly copying the code from expire_register, just use the function
that "does the right thing".
(closes issue #16033)
Reported by: kkm
Patches:
016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
Tested by: kkm, tilghman, twilson
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Because crashing is bad.
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r315644 | twilson | 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) | 32 lines
Merged revisions 315643 via svnmerge from
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r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines
Merged revisions 315596 via svnmerge from
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r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines
Allow transfer loops without allowing forwarding loops
We try to avoid the situation where two phones may be forwarded to each other
causing an infinite loop by storing each dialed interface in a channel
datastore and checking the list before dialing out. This works, but currently
breaks situations like A calls B, A transfers B to C, B transfers C to A, and A
transfers C to B. Since human interaction is happening here and not an
automated forwarding loop, it should be allowed.
This patch removes the dialed_interfaces datastore when a call is bridged (a
suggestion from the brilliant mmichelson). If a call is being bridged, it
should be safe to assume that we aren't stuck in a loop.
Since we are now handling this is the bridge code, the previous attempts at
handling it in app_dial and app_queue are removed.
Review: https://reviewboard.asterisk.org/r/1195/
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r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) | 21 lines
The 'e' special extension fails to trigger in at least two cases.
The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if
any of them do not exist. Many of the places the 'e' extension was
supposed to be invoked fail because the priority was set wrong. There
were two places where the 'e' extension was not even checked for fall
back.
* Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T'
extension check and added the 'e' extension as a fall back to the two
missing locations.
* Prioritized and optimized some hangup tests associated with the 'e'
extension.
(closes issue #19136)
Reported by: kshumard
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1196/
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r315503 | tilghman | 2011-04-26 14:32:50 -0500 (Tue, 26 Apr 2011) | 28 lines
Merged revisions 315502 via svnmerge from
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r315502 | tilghman | 2011-04-26 14:22:52 -0500 (Tue, 26 Apr 2011) | 21 lines
Merged revisions 315501 via svnmerge from
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r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines
Fix the bounds-checking code.
The code that set the bit within the select bitfield was correct, but the
bounds-checking code was not. The change to that line uses the new _bitsize
macro for clarity. Also, FD_ZERO macro did not zero-out anything but the
first word of the bitfield, so this could have caused problems with modules
using that macro with the expanded bitfield.
(closes issue #18773)
Reported by: jamicque
Patches:
20110423__issue18773.diff.txt uploaded by tilghman (license 14)
Tested by: chris-mac
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r315452 | rmudgett | 2011-04-26 13:00:34 -0500 (Tue, 26 Apr 2011) | 1 line
Add missing set of name valid flag when dialing.
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r315446 | russell | 2011-04-26 12:40:23 -0500 (Tue, 26 Apr 2011) | 14 lines
chan_local: resolve a deadlock.
This patch resolves a fairly complex deadlock that can occur with the
combination of chan_local and a dialplan switch, such as dynamic realtime
extensions, which pulls autoservice into the picture when doing a dialplan
lookup.
(closes issue #18818)
Reported by: nic
Patches:
issue18818.patch uploaded by jthurman (license 614)
18818.v1.txt uploaded by russell (license 2)
Tested by: nic, jthurman, kterzi, steve-howes, sysreq, IshMalik
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r315394 | pabelanger | 2011-04-25 22:18:50 -0400 (Mon, 25 Apr 2011) | 14 lines
Merged revisions 315393 via svnmerge from
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r315393 | pabelanger | 2011-04-25 22:17:43 -0400 (Mon, 25 Apr 2011) | 7 lines
Add back CLI command 'dialplan save'
(closes issue #19140)
Reported by: lmadsen
Patches:
__20110419_dialplan_save.patch.txt uploaded by lmadsen (license 10)
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r315349 | rmudgett | 2011-04-25 16:49:00 -0500 (Mon, 25 Apr 2011) | 9 lines
When using MGCP realtime gateway definitions, random crashes occur.
Fixed incorrect linked list node removal for realtime gateways.
(closes issue #18291)
Reported by: nahuelgreco
Patches:
dangling-pointers-when-pruning.patch uploaded by nahuelgreco (license 162)
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r315259 | russell | 2011-04-25 14:37:32 -0500 (Mon, 25 Apr 2011) | 24 lines
Merged revisions 315258 via svnmerge from
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r315258 | russell | 2011-04-25 14:31:44 -0500 (Mon, 25 Apr 2011) | 17 lines
Merged revisions 315257 via svnmerge from
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r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011) | 10 lines
Be more flexible with unknown chunks in wav files.
This patch makes format_wav ignore unknown chunks instead of erroring
out on them.
(closes issue #18306)
Reported by: jhirsch
Patches:
wav_skip_unknown_blocks.diff uploaded by jhirsch (license 1156)
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r315213 | russell | 2011-04-25 14:04:28 -0500 (Mon, 25 Apr 2011) | 14 lines
Merged revisions 315212 via svnmerge from
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r315212 | russell | 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011) | 7 lines
Don't link non-cached realtime peers into the peers_by_ip container.
(closes issue #18924)
Reported by: wdoekes
Patches:
issue18924_uncached_realtime_peers_leak-1.6.2.17.patch uploaded by wdoekes (license 717)
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r315053 | alecdavis | 2011-04-25 19:14:32 +1200 (Mon, 25 Apr 2011) | 23 lines
Merged revisions 315052 via svnmerge from
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r315052 | alecdavis | 2011-04-25 19:11:12 +1200 (Mon, 25 Apr 2011) | 16 lines
Merged revisions 315051 via svnmerge from
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r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr 2011) | 11 lines
chan_local:check_bridge() misplaced misplaced ast_mutex_unlock
if !p->chan->_bridge->_softhangup path isn't followed, brigde remains locked.
(closes issue #19176)
Reported by: alecdavis
Patches:
bug19176.diff.txt uploaded by alecdavis (license 585)
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r315001 | alecdavis | 2011-04-23 10:59:18 +1200 (Sat, 23 Apr 2011) | 12 lines
chan_dahdi: Can't return to normal ring after distinctive ring on FXS
clear a previous distinctivering pattern before each new call
(closes issue #18985)
Reported by: bromont
Patches:
bug18985.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, bromont
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r314959 | mnicholson | 2011-04-22 16:20:08 -0500 (Fri, 22 Apr 2011) | 24 lines
Merged revisions 314958 via svnmerge from
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r314958 | mnicholson | 2011-04-22 15:49:45 -0500 (Fri, 22 Apr 2011) | 17 lines
Merged revisions 311203,314908 via svnmerge from
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r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar 2011) | 4 lines
Don't hold the pvt lock while streaming a file.
ABE-2756
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r314908 | mnicholson | 2011-04-22 15:01:48 -0500 (Fri, 22 Apr 2011) | 4 lines
Prevent the login thread and the app threads from using the asterisk channel at the same time.
ABE-2756
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r314779 | tzafrir | 2011-04-22 16:59:43 +0300 (ו', 22 אפר 2011) | 2 lines
Fix a few typos (shown by Lintian)
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r314780 | russell | 2011-04-22 09:02:23 -0500 (Fri, 22 Apr 2011) | 18 lines
Merged revisions 314778 via svnmerge from
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r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011) | 11 lines
Initialize buffers in getvar and getvarfull.
Initialize the buffers used to hold the result from GET VARIABLE or
GET VARIABLE FULL. The bug report shows func_read returning garbage in
the result. It assumed that the buffer passed in was initialized, like many
other functions do. In the more common code path (through the dialplan), it
is initialized, so just initialize it here too.
(closes issue #19050)
Reported by: johnz
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PRIShowSpans works like the AMI action DAHDIShowChannels but for PRI
spans. It is similar to the CLI command "pri show spans".
(closes issue #15980)
Reported by: dwery
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r314732 | rmudgett | 2011-04-21 17:38:44 -0500 (Thu, 21 Apr 2011) | 1 line
Correct DAHDIShowChannels XML documentation.
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r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines
Merged revisions 314620 via svnmerge from
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r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines
Merged revisions 314607 via svnmerge from
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r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines
Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so.
Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action.
AST-2011-005
AST-2011-006
(closes issue #18787)
Reported by: kobaz
(related to issue #18996)
Reported by: tzafrir
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Includes a new highly optimized and customizable
ConfBridge application capable of mixing audio at
sample rates ranging from 8khz-192khz.
Review: https://reviewboard.asterisk.org/r/1147/
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r314550 | twilson | 2011-04-20 17:23:04 -0700 (Wed, 20 Apr 2011) | 13 lines
Merged revisions 314549 via svnmerge from
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r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) | 6 lines
Don't allocate more space than necessary for a sip_pkt
This extra allocation is a hold-over from when pkt->data was a
character array. Now that it is an allocated string, just allocate
enough for the sip_pkt.
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Review: https://reviewboard.asterisk.org/r/1157/
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Previously, the DAHDI format bit fields matched up with the Asterisk
bitfields. Since the Asterisk codec bit fields were replaced in r306010,
codec_dahdi needs to contain the formats itself. In the future, the DAHDI
formats should either change to something other than bitfields, or the
bitfields need to move from include/dahdi/kernel.h to
include/dahdi/user.h.
Signed-off-by: Shaun Ruffell <sruffell@digium.com>
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r314417 | rmudgett | 2011-04-20 11:54:02 -0500 (Wed, 20 Apr 2011) | 1 line
AST_CONTROL_XXX comment changes.
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not allwaya accurate.
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r314358 | twilson | 2011-04-19 22:25:15 -0700 (Tue, 19 Apr 2011) | 4 lines
Initialize track pointer
ast_reentrancy_init checks to see if it is NULL before initializing with calloc
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r314251 | lmadsen | 2011-04-19 10:42:10 -0500 (Tue, 19 Apr 2011) | 8 lines
Use SSLv23_client_method instead of old SSLv2 only.
(closes issue #19095)
(closes issue #19138)
Reported by: tzafrir
Patches:
no_ssl2.diff uploaded by tzafrir (license 46)
Tested by: russell, chazzam
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r314206 | lmadsen | 2011-04-19 09:28:15 -0500 (Tue, 19 Apr 2011) | 14 lines
Merged revisions 314205 via svnmerge from
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r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19 Apr 2011) | 6 lines
Remove duplicate documentation from func_channel.c
(closes issue #18970)
Reported by: IgorG
Patches:
func_channel.c.doc.diff uploaded by IgorG (license 20)
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r314203 | lmadsen | 2011-04-19 09:24:25 -0500 (Tue, 19 Apr 2011) | 15 lines
Merged revisions 314202 via svnmerge from
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r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011) | 7 lines
Update seconds to milliseconds in ast_verb output.
(closes issue #19084)
Reported by: smurfix
Patches:
app_dial.patch uploaded by smurfix (license 547)
Tested by: lmadsen, smurfix
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message to asterisk-dev)
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The "controlling user number" is always the number of the voice mail box
which is identical with the subscriber number itself. This number which
is listed in the ISDN phone MWI menu cannot be called back to contact the
voice mail box. The controlling user number should be made configurable.
JIRA ABE-2738
JIRA SWP-2846
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r314069 | rmudgett | 2011-04-18 11:10:10 -0500 (Mon, 18 Apr 2011) | 22 lines
The AsyncAGI command loop is lax in the value it returns for the return status.
* Return correct status: SUCCESS/FAILED/HANGUP. Previously, abnormal
exits from the command loop such as hangup would return SUCCESS.
* The "asyncagi break" command now returns SUCCESS and is now the only way
to break the command loop with that status. Previously, it returned
FAILED.
* The AMI event AsyncAGI End is no longer sent if the AsyncAGI Start event
is not sent. Previously, this happened because of an error setting up the
AGI pipes.
* All executed AGI commands now get an AsyncAGI Exec result event.
Previously, if the command returned failure (because of hangup), the
command loop just exited with FAILURE and did not send the AsyncAGI Exec
result event.
* Makes sure that the channel frame queue is empty on hangup.
Review: https://reviewboard.asterisk.org/r/1183/
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r314068 | rmudgett | 2011-04-18 11:02:12 -0500 (Mon, 18 Apr 2011) | 7 lines
Unclear code in app_dial.c.
Make code formatting clear.
(closes issue #19134)
Reported by: oej
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