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kFreeBSD is GNU (with glibc) on to of a FreeBSD kernel. See
http://glibc-bsd.alioth.debian.org/porting/PORTING
This patch gets Asterisk close to building on Debian kFreeBSD i386,
mainly by adding an extra test for __GLIBC__ in one or two (or more)
places.
OSARCH is set to 'kfreebsd-gnu'
DAHDI support (and support for chan_vpb) was not tested.
Review: https://reviewboard.asterisk.org/r/858/
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r282366 | tilghman | 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010) | 4 lines
Fix our FRACKing issue with chan_iax2 a different way.
Review: https://reviewboard.asterisk.org/r/861/
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r282334 | rmudgett | 2010-08-13 18:53:36 -0500 (Fri, 13 Aug 2010) | 6 lines
PRI CCSS may use a stale dial string for the recall dial string.
If an outgoing call negotiates a different B channel than initially
requested, the saved original dial string was not transferred to the new B
channel. CCSS uses that dial string to generate the recall dial string.
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r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines
remove current STUN support from chan_sip.c
This patch removes the current broken/useless stun
support from chan_sip.
(closes issue #17622)
Reported by: philipp2
Review: https://reviewboard.asterisk.org/r/855/
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r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010) | 2 lines
res_stun_monitor and corresponding options CHANGES documentation
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r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) | 4 lines
res_stun_monitor for monitoring network changes behind a NAT device
Review: https://reviewboard.asterisk.org/r/854
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r282236 | dvossel | 2010-08-13 13:58:10 -0500 (Fri, 13 Aug 2010) | 23 lines
Merged revisions 282235 via svnmerge from
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r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) | 16 lines
only do magic pickup when notifycid is enabled
A new way of doing BLF pickup was introduced into 1.6.2. This feature
adds a call-id value into the XML of a SIP_NOTIFY message sent to alert
a subscriber that a device is ringing. This option should only be enabled
when the new 'notifycid' option is set... but this was not the case. Instead
the call-id value was included for every RINGING Notify message, which
caused a regression for people who used other methods for call pickup.
(closes issue #17633)
Reported by: urosh
Patches:
chan_sip.txt uploaded by urosh (license )
blf_cid_issue.diff uploaded by dvossel (license 671)
Tested by: dvossel, urosh, okrief, alecdavis
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r282200 | twilson | 2010-08-13 11:00:02 -0500 (Fri, 13 Aug 2010) | 10 lines
Detect when libsrtp cannot be linked in a shared library
The libsrtp build system currently does not produce a shared library
or a static library compiled with -fPIC, so on 64-bit systems it is
possible that we will get a compile error if libsrtp is installed and
res_srtp is selected in menuselect.
This patch attempts to detect this situation and provide the user with
instructions to work around the problem.
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r282201 | twilson | 2010-08-13 11:02:20 -0500 (Fri, 13 Aug 2010) | 2 lines
Whitespace fix :-/
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r282131 | qwell | 2010-08-12 17:51:44 -0500 (Thu, 12 Aug 2010) | 16 lines
Merged revisions 282130 via svnmerge from
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r282130 | qwell | 2010-08-12 17:50:54 -0500 (Thu, 12 Aug 2010) | 9 lines
Merged revisions 282129 via svnmerge from
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r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) | 1 line
Register CLI commands before parsing config, in case there is a config error.
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r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) | 7 lines
Separate call completion config parameter allocation and default initialization.
If you ever have a need to reset the call completion config parameters
to defaults, now you can.
And no Virginia, C++ idioms do not always work in C.
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r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) | 4 lines
Add a "core reload" CLI command.
Review: https://reviewboard.asterisk.org/r/859/
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r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines
improved translation paths for wideband codecs
The problem I'm addressing is that Asterisk's current
method of building the least cost translation paths
between codecs does not take into account sample rate.
For instance, it was possible for siren14 (a 32khz codec),
to contain the a translation path to siren7 (a 16khz
audio codec) that goes through slin at 8khz. In this
case Asterisk takes a 32khz codec, down samples it to
8khz and then up samples it to 16khz which is terrible
regardless if it is computationally less expensive. This
patch now builds translation paths that give priority to
maintaining the best possible sample rate before taking
into consideration computational cost. This patch also
adds cli commands to expose what translation paths are
actually being used.
Changes:
1. Translation paths will never contain a step that changes
the sample rate unless absolutely necessary.
2. When choosing the best codec to make two channels compatible.
Shared codecs with the highest sample rate are given priority.
3. A new cli command to show all translation paths available
for a specific codec 'core show translation paths [codec name]'
has been added.
4. 'core show translation' which displays the translation
matrix now includes the new higher bit audio codecs in the table.
5. 'core show channel [channel name]' now displays the
translation paths if translation is used.
(closes issue #16841)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/842/
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r282015 | russell | 2010-08-12 13:03:56 -0500 (Thu, 12 Aug 2010) | 2 lines
Put back pointer value output for ast_debug(), such that it is only removed for verbose output.
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r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010) | 5 lines
Remove debugging output from verbose messages.
Pointer values to internal objects is not terribly useful to users in the
verbose messages about adding extensions and contexts.
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r281913 | jpeeler | 2010-08-11 22:03:37 -0500 (Wed, 11 Aug 2010) | 34 lines
Merged revisions 281912 via svnmerge from
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r281912 | jpeeler | 2010-08-11 22:01:38 -0500 (Wed, 11 Aug 2010) | 27 lines
Merged revisions 281911 via svnmerge from
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r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) | 20 lines
Ensure SSRC is changed when media source is changed to resolve audio delay.
This change causes the SSRC to change right before the channels are bridged,
which is what used to happen. It seems that fixes were made to attempt limiting
SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC
with this change.
There are two other control frames sent in ast_channel_bridge that probably
should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave
this change up to the discretion of resolving issue #17007.
For reference - old review implementing new control frame SRCCHANGE:
https://reviewboard.asterisk.org/r/540
(closes issue #17404)
Reported by: sdolloff
Patches:
bug17404.patch uploaded by jpeeler (license 325)
Tested by: sdolloff
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r281875 | lmadsen | 2010-08-11 16:12:13 -0500 (Wed, 11 Aug 2010) | 21 lines
Merged revisions 281873 via svnmerge from
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r281873 | lmadsen | 2010-08-11 16:09:47 -0500 (Wed, 11 Aug 2010) | 14 lines
Merged revisions 281819 via svnmerge from
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r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010) | 6 lines
Add Danish support to say.conf.sample
(closes issue #17836)
Reported by: RoadKill
Patches:
say.conf.sample.patch.dk uploaded by RoadKill (license 933)
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r281874 | mnicholson | 2010-08-11 16:11:54 -0500 (Wed, 11 Aug 2010) | 10 lines
handle all possible responses to REFER requests
(closes issue #17486)
Reported by: davidw
Patches:
Issue17486-counterbid.diff.txt uploaded by davidw (license 780)
Tested by: davidw
Review: https://reviewboard.asterisk.org/r/837/
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r281870 | rmudgett | 2010-08-11 15:30:29 -0500 (Wed, 11 Aug 2010) | 4 lines
Fix a call to analog_set_pulsedial() not setting 0 or 1 only.
* Also a couple minor tweaks.
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r281764 | lmadsen | 2010-08-11 12:54:56 -0500 (Wed, 11 Aug 2010) | 21 lines
Merged revisions 281763 via svnmerge from
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r281763 | lmadsen | 2010-08-11 12:54:09 -0500 (Wed, 11 Aug 2010) | 14 lines
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r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010) | 6 lines
Allow say.conf to handle large numbers ending with multiple zeros.
(closes issue #17833)
Reported by: RoadKill
Patches:
say.conf.sample.patch.largenumbers uploaded by RoadKill (license 933)
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r281760 | mnicholson | 2010-08-11 12:27:59 -0500 (Wed, 11 Aug 2010) | 4 lines
Avoid a deadlock in add_header_max_forwards().
Related to r276951
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r281723 | tilghman | 2010-08-11 10:18:40 -0500 (Wed, 11 Aug 2010) | 14 lines
Merged revisions 281722 via svnmerge from
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r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11 Aug 2010) | 7 lines
Only set status TIMEOUT, if we have no digits.
(closes issue #15188)
Reported by: jcovert
Patches:
app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license 551)
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r281687 | simon.perreault | 2010-08-11 09:30:59 -0400 (Wed, 11 Aug 2010) | 9 lines
Fix parsing of IPv6 address literals in outboundproxy
(closes issue #17757)
Reported by: oej
Patches:
17757.diff uploaded by sperreault (license 252)
sip.conf.diff uploaded by sperreault (license 252)
Tested by: oej
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r281650 | russell | 2010-08-10 16:47:31 -0500 (Tue, 10 Aug 2010) | 5 lines
Change the default value for alwaysauthreject in sip.conf to "yes".
(closes issue #17756)
Reported by: oej
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r281575 | russell | 2010-08-10 13:05:07 -0500 (Tue, 10 Aug 2010) | 16 lines
Merged revisions 281574 via svnmerge from
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r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010) | 9 lines
Don't move the time threshold for running scheduled events on every iteration.
Instead, only calculate the time threshold each time ast_sched_runq() is called.
(closes issue #17742)
Reported by: schmidts
Patches:
sched.c.patch uploaded by schmidts (license 1077)
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r281568 | russell | 2010-08-10 12:48:42 -0500 (Tue, 10 Aug 2010) | 22 lines
Merged revisions 281567 via svnmerge from
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r281567 | russell | 2010-08-10 12:47:13 -0500 (Tue, 10 Aug 2010) | 15 lines
Merged revisions 281566 via svnmerge from
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r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) | 8 lines
Reset visible indication after answer.
(closes issue #17641)
Reported by: klaus3000
Patches:
ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65)
Tested by: schmidts
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r281532 | russell | 2010-08-10 11:54:20 -0500 (Tue, 10 Aug 2010) | 8 lines
Ensure that the proper external address is used for the RTP destination.
(closes issue #17044)
Reported by: ebroad
Tested by: ebroad
Review: https://reviewboard.asterisk.org/r/566/
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r281529 | russell | 2010-08-10 11:21:58 -0500 (Tue, 10 Aug 2010) | 8 lines
Resolve a problem with channel name tab completion.
Hitting tab without typing any part of a channel name resulted in no results.
This now results in getting a full list of active channels, just as it did
in previous versions of Asterisk.
Review: https://reviewboard.asterisk.org/r/818/
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r281466 | jpeeler | 2010-08-09 18:04:02 -0500 (Mon, 09 Aug 2010) | 2 lines
Add some more stuff to copy from 281429.
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r281432 | dvossel | 2010-08-09 15:47:53 -0500 (Mon, 09 Aug 2010) | 20 lines
Merged revisions 281430 via svnmerge from
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r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010) | 13 lines
fixes SIP peers memory leak
We zeroed out the peer's addr before it was removed from the
peers_by_ip container. This made it impossible to be removed
from the container as the addr is the key used by the container
to find the peer.
(closes issue #17774)
Reported by: kkm
Patches:
017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
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r281429 | jpeeler | 2010-08-09 15:43:54 -0500 (Mon, 09 Aug 2010) | 27 lines
Merged revisions 281391 via svnmerge from
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r281391 | jpeeler | 2010-08-09 15:07:29 -0500 (Mon, 09 Aug 2010) | 20 lines
Merged revisions 281390 via svnmerge from
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r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) | 13 lines
Prevent loss of Caller ID information set on local channel after masquerade.
Caller ID set on the channel before a masquerade occurs when using a local
channel would cause the information to be lost. The problem was that the
information was set on a channel destined to be hung up. The somewhat confusing
fix is to detect if any Caller ID has been set on the channel and if so
preswap the Caller ID data so that basically the masquerade puts the data back.
(closes issue #17138)
Reported by: kobaz
Review: https://reviewboard.asterisk.org/r/847/
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r281358 | mnicholson | 2010-08-09 09:49:38 -0500 (Mon, 09 Aug 2010) | 4 lines
Validate minrate, maxrate, and modem settings before attempting a fax session.
FAX-224
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r281356 | simon.perreault | 2010-08-09 10:31:40 -0400 (Mon, 09 Aug 2010) | 2 lines
Added comment about IPv4-mapped IPv6 addresses and the output of netstat.
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r281325 | russell | 2010-08-09 07:51:43 -0500 (Mon, 09 Aug 2010) | 2 lines
Add a couple of default values to the documentation of cdr.conf.
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r281294 | russell | 2010-08-09 07:14:34 -0500 (Mon, 09 Aug 2010) | 5 lines
Reorder some options in cdr.conf.sample.
Put all of the options that affect the contents of CDRs together, instead
of having the batch mode options in the middle of them.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Cleaned up handling of onhook indications and added indications if more than one sub on device. Also fixes issue in 12324 so that the phone can call itself without locking up.
(closes issue #17692)
Reported by: jmhunter
Patches:
chan_skinny-transfer-v4.txt uploaded by DEA (license 3)
skinnytransfver.v8.diff uploaded by wedhorn (license 30)
Tested by: jmhunter, salecha, wedhorn
Review: NA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Move call answering stuff into new setsubstate_connected. Also add sub->substate var and set it to SUBSTATE_CONNECTED in setsubstate_connected.
(closes issue #17772)
Reported by: wedhorn
Patches:
cleanup.stateconnected2.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, salecha
Review: NA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #17770)
Reported by: salecha
Patches:
skinny.answercrash.diff uploaded by wedhorn (license 30)
Tested by: salecha
Review: NA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281085 | tilghman | 2010-08-06 13:57:10 -0500 (Fri, 06 Aug 2010) | 8 lines
Fix alignment of stringfields on the SPARC architecture
(closes issue #17789)
Reported by: Ian Mason
Patches:
20100806__issue17789__2.diff.txt uploaded by tilghman (license 14)
Tested by: Ian_Mason
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r281052 | russell | 2010-08-05 08:16:11 -0500 (Thu, 05 Aug 2010) | 16 lines
Merged revisions 281051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010) | 9 lines
Cleanup default option value handling for cdr.conf [general].
The default values would differ depending on whether or not cdr.conf exists.
That is no longer the case.
Apply a default value to the unanswered option.
Define all default values as named constants.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280984 | tilghman | 2010-08-05 02:46:36 -0500 (Thu, 05 Aug 2010) | 22 lines
Merged revisions 280983 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r280983 | tilghman | 2010-08-05 02:40:47 -0500 (Thu, 05 Aug 2010) | 15 lines
Merged revisions 280982 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) | 8 lines
Change context lock back to a mutex, because functionality depends upon the lock being recursive.
(closes issue #17643)
Reported by: zerohalo
Patches:
20100726__issue17643.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r280909 | mnicholson | 2010-08-04 10:11:13 -0500 (Wed, 04 Aug 2010) | 2 lines
Initialize FAXOPT() status variables in sendfax and receivefax instead of when the details structure is created.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280879 | tilghman | 2010-08-04 09:04:07 -0500 (Wed, 04 Aug 2010) | 14 lines
Check cur value before attempting a deref.
(closes issue #17775)
Reported by: svinson
Patches:
20100804__issue17775.diff.txt uploaded by tilghman (license 14)
Tested by: svinson
(closes issue #17743)
Reported by: tgruenberg
Patches:
20100804__issue17775.diff.txt uploaded by tilghman (license 14)
Tested by: tgruenberg
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280809 | tilghman | 2010-08-03 15:25:10 -0500 (Tue, 03 Aug 2010) | 12 lines
Sneak FIELDNUM() into 1.8. Returns a 1-based index into a list of a specified item.
Matches up with FIELDQTY() and CUT().
(closes issue #17713)
Reported by: gareth
Patches:
svn-279754.diff uploaded by gareth (license 208)
Tested by: gareth, tilghman
Review: https://reviewboard.asterisk.org/r/810/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r280778 | simon.perreault | 2010-08-03 15:54:03 -0400 (Tue, 03 Aug 2010) | 9 lines
Fixed IPv6-related SIP parsing bugs.
(closes issue #17663)
Reported by: oej
Patches:
diff uploaded by sperreault (license 252)
diff2 uploaded by sperreault (license 252)
get_domain.diff uploaded by sperreault (license 252)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r280777 | simon.perreault | 2010-08-03 15:53:07 -0400 (Tue, 03 Aug 2010) | 8 lines
Better documentation related to IPv6.
(closes issue #17737)
Reported by: oej
Patches:
doc.diff uploaded by sperreault (license 252)
Tested by: mmichelson
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properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280742 | russell | 2010-08-03 13:48:45 -0500 (Tue, 03 Aug 2010) | 4 lines
Remove the MP3 decoder source code and replace it with a small shell script.
Review: https://reviewboard.asterisk.org/r/836/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r280740 | tilghman | 2010-08-03 13:42:24 -0500 (Tue, 03 Aug 2010) | 9 lines
Merged revisions 280739 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010) | 2 lines
Document -B and -W flags and regenerate manpage from sgml
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #17663)
Reported by: oej
Patches:
diff uploaded by sperreault (license 252)
diff2 uploaded by sperreault (license 252)
get_domain.diff uploaded by sperreault (license 252)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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