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Also, add some documentation supporting the use of astcanary.
(closes issue #14538)
Reported by: KNK
Patches:
asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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what kind of encryption is enabled as well.
Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not.
(closes issue #14427)
Reported by: snuffy
Patches:
iax_show_trunks.diff uploaded by snuffy (license 35)
2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, dvossel, snuffy
Review: http://reviewboard.digium.com/r/173/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines
Skip check for extension when subscribing for MWI.
Since the remote side is not actually subscribing to a specific extension when
subscribing for MWI just skip the check to see if the extension exists. They can't use it
to specify the mailbox either since we require configuration of that in sip.conf
(closes issue #14531)
Reported by: festr
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines
Fix infinite DTMF when a BEGIN is received without an END.
This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem. The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.
In passing, I removed the dtmfsamples variable which was completed unused. I
also removed a redundant setting of the lastrxts variable.
(closes issue #14460)
Reported by: moliveras
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #14372)
Reported by: fhackenberger
Patches:
voicemail_individual_subject_and_body_1.6.1 uploaded by fhackenberger (license 592)
with additional fixes by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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chan_sip's headers. requested by oej.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers". Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled.
(closes issue #14523)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #14522)
Reported by: pj
Tested by: russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Added:
SKINNYdevices
SKINNYshowdevice
SKINNYlines
SKINNYshowline
(closes issue #14521)
Reported by: mvanbaak
Review: http://reviewboard.digium.com/r/170/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #14513)
Reported by: snuffy
Patches:
bug14513_fixsysinfo.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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hashtab documentation a bit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #14512)
Reported by: snuffy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When running asterisk as non-root and without this patch the pidfile wants
to go into /var/run/asterisk.pid. This directory is not writable for
the non-root user and changing permissions is not an option.
Putting it in /var/run/asterisk/asterisk.pid makes it possible
to set permissions on the /var/run/asterisk dir so everything
works as it should be.
Patched committed is based on pabelanger's patch.
(closes issue #13153)
Reported by: pabelanger
Patches:
2009012900_bug13153-nonrootscripts.diff.txt uploaded by mvanbaak (license 7)
Review: http://reviewboard.digium.com/r/139/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines
Don't print the CR-NL combination when we aren't outputting to the manager.
An embedded CR-NL in a CLI command screws up several AMI parsers that don't
expect to see that combination in the middle of output.
(Closes issue #14305)
Reported by: martins
Patch by: tilghman
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines
This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed.
Fixed for snuff-home on -dev channel.
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Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred
integer to indicate the number of pages transferred (so far) during the fax
session. The spandsp-0.0.6pre4 release removed the pages_transferred integer
and replaced it with two different integers - pages_tx and pages_rx. This
revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards
compatibility for previous spandsp releases.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r177696 | dvossel | 2009-02-20 14:17:37 -0600 (Fri, 20 Feb 2009) | 8 lines
Fixes issue with undefined audio codecs in chan_iax2
During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-15 are not defined, these bits are never turned off. In trunk, bits 13-15 are defined, which means 1.4 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities.
(closes issue #14283)
Reported by: jcovert
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(closes issue #14231)
Reported by: jcovert
Patches:
20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14)
corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551)
Tested by: jcovert
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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in copy_request
(issue #14478)
Reported by: erik_dedecker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
Trunk was already pretty 8-bit clean; but I'm still
removing the --full from the flex command so everything
is uniform.
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r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
This patch fixes a problem with 8-bit input to the ast_expr2 scanner.
The real culprit was the --full argument to flex
in the Makefile! This causes a 7-bit scanner to be
generated.
I reviewed the rules and found one rule where I needed
to specifically include 8-bit chars for a token.
I tested against the text supplied by ibercom, and
all looks very well.
This has been there a surprisingly long time!
(closes issue #14498)
Reported by: ibercom
Patches:
14498.patch uploaded by murf (license 17)
Tested by: murf
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) | 7 lines
Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads.
(closes issue #14253)
Reported by: Skavin
Patches:
20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
Tested by: Skavin
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trunk. Will do some
testing to make sure this works properly, but from reading the code, it does seem to work
as it should.
Blocked revisions 177450 via svnmerge
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r177450 | oej | 2009-02-19 19:58:57 +0100 (Tor, 19 Feb 2009) | 2 lines
Force a MWI notification after subscribe request. Reported by the Resiprocate dev team. Thanks!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines
If we are able to create a speech structure unset the ERROR variable in case it was previously set.
(issue #LUMENVOX-13)
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asterisk-dev mailing list. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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time ago.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | 34 lines
This patch fixes a regression of sorts that was introduced in
rev 24425.
It basically fixes AST-190/ABE-1782.
What was wrong: the user has 6000 extensions in one context; and
then 6000 contexts, one per extension. The parser could only handle
about 4893 of the 6000 extens in the single context.
This was due to the regression I mentioned. To get rid of
shift/reduce conflicts, Luigi set up right-recursive lists
for globals, context elements, switch lists, and statements.
Right recursive lists got rid of the warnings, but instead, they
use up a tremendous amount of stack space when the lists are long.
I saw this a few years back, and resolved not to fix it until
someone complained. That day has arrived!
After the changes were made, I ran the regression test suite,
and there were no problems.
I took the test case the user provided, and added 100,000
extensions to the single context, that already had 6,000 extens
in it. (I'll see your 6, and raise you 100!) It takes a few minutes
to read it all in, check it and generate code for it, but no
problems.
So, I think I can say that fundamentally, there are no longer
any limits on the number of items you can place in contexts,
statement blocks, switches, or globals, beyond your virt mem
constraints.
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ensure that the samples are byte-swapped to network order if needed. also, when a smoother is operating on a format that has a sample rate other than 8000 samples per second, use the proper sample rate for computing delivery timestamps.
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action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock. Instead, they should hold the lock until the masquerade is complete. This will guarantee the channel remains and prevent any other weirdness from occurring. In action_bridge() some more weirdness comes into play. Both channels are needlessly locked at the same time and perform the exact same logic. It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up. There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it.
issue# 14296
Review: http://reviewboard.digium.com/r/167/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Several changes in PTLib have occurred requiring build time detection. Changes
accounted for include the library name change, config option change, install
location change, and a boolean type change which is handled by ast_ptlib.h.
Also, the sed check has been modified to properly work with autoconf >= 2.62.
(closes issue #14224)
Reported by: bergolth
Patches:
asterisk-autoconf-sed.patch uploaded by bergolth (license 661)
asterisk-pwlib-v3.patch uploaded by bergolth (license 661)
Tested by: jpeeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r177160 | jpeeler | 2009-02-18 14:06:45 -0600 (Wed, 18 Feb 2009) | 15 lines
Modify h323 to build against PTLib as well as the older PWLib
Several changes in PTLib have occurred requiring build time detection. Changes
accounted for include the library name change, config option change, install
location change, and a boolean type change which is handled by ast_ptlib.h.
Also, the sed check has been modified to properly work with autoconf >= 2.62.
(closes issue #14224)
Reported by: bergolth
Patches:
asterisk-autoconf-sed.patch uploaded by bergolth (license 661)
asterisk-pwlib-v3.patch uploaded by bergolth (license 661)
Tested by: jpeeler
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Enabling this option by default proved to be a bad idea, as the talker detection
is not very reliable. So, make it optional again, and off by default.
(issue #13801)
Reported by: justdave
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) | 2 lines
Document the return value of the update method (as requested on -dev list)
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(closes issue #14497)
Reported by: vinsik
Patches:
chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)
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the amount available.
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hints are present.
Reason: when I re-engineered the merge_and_delete func to
reduce its lock time, I failed to notice that the
functions it calls still also do locking as before.
This leads to deadlocks on dialplan reloads, when
there are actually living, subscribed hints registered
in the system.
While the reporter come across this problem while using
AEL, I might note that these deadlocks should also happen
if extensions.conf were used.
Here I added these routines to pbx.c:
ast_add_extension_nolock
add_pri_lockopt
ast_add_extension2_lockopt
find_context
add_hint_nolock
All of the above routines are static and restricted
to be used only within pbx.c, and more specifically
within the merge_contexts_and_delete routine.
They are pretty much the same as their counterparts
except they don't lock contexts or hints.
Most of them now do the real work of their
name-alike, with optional locking via extra arguments,
and are called by their name-alike. The goal was to
have the original functions so they would behave
exactly as before.
Both PJ and I tested these fixes, and the deadlocking
problem is no longer encountered.
(closes issue #14357)
Reported by: pj
Patches:
14357.diff uploaded by murf (license 17)
Tested by: pj, murf
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The big problem here is that the 3rd argument provided in these uses of strncpy()
did not reserve a byte for the null terminator, leaving the potential for writing
one byte past the end of the buffer.
Aside from this, there were coding guidelines violations with regards to spacing,
as well as hard coded lengths being used instead of sizeof().
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The previous implementation of T38 faxdetect resulted in both sides of the
call jumping to a fax extension when both sides had 't38pt_udptl=yes' and
'faxdetect=yes' in sip.conf and a 'fax' extension in the current context.
This revision will jump to a 'fax' extension on incoming calls only.
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This commit brings in the changes that were living out on the
svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch. codec_dahdi.c now
always uses signed linear as the simple codec so that a soft g729 codec will
not end up being preferred to the hardware codec. There are also changes to
allow codec_dahdi.c to feed packets to the hardware in the native sample size of
the codec. This solves problems with choppy audio when using G723.
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