The installation script and the new configure option --with-pjproject-bundled
aimed to accomplish the same. However, the installation script was out of
date. Users should go for the maintained configure option, or the Wiki.
ASTERISK-24598
Change-Id: Icbf4b562f81f7c05bd24a3805bd46c0beb4ebd44
This re-enables the script ./contrib/scripts/install_prereq on Fedora 22 and
newer, and on RHEL/CentOS when the option strict=1 was set for yum install.
ASTERISK-27598
Reported by: Hunter Stevens, Said Masoud
Change-Id: I40f9517122aaa6906e8fc0962b4b8008dfddb368
The OUTPUTDIR environment variable can now be set either in the
environment itself or in ast_debug_tools.conf. If set, it's used
for all work products instead of /tmp.
Also added the --tarball-config option that includes the contents
of /etc/asterisk when either --tarball-coredumps or --tarball-results
are used.
Change-Id: I66b2553319df61caea5b313d084f51978f730b4c
Since Asterisk 13.17, libSRTP 2.x is supported. Therefore, its latest version
is installed again via the script install_prereq.
ASTERISK-27356
Change-Id: I13125839a79052356469e41edacbebff0a937d39
Adds an extra option, --asterisk-bin=<path> to ast_coredumper. If
provided, the binary given to gdb will be the parameter, rather than
asterisk from the PATH.
ASTERISK-27380 #close
Change-Id: I25f5b91eb75059b0fb2f142e468c26b283b0a9f3
The --tarball-coredump option now creates a gzipped tarball of
coredumps processed, their results txt files and copies of
/etc/os-release, /usr/sbin/asterisk, /usr/lib(64)/libasterisk* and
/usr/lib(64)/asterisk as those files are needed to properly examine
the coredump. The file will be named
/tmp/asterisk.<timestamp>.coredumps.tar.gz or
/tmp/asterisk-<uniqueid>.coredumps.tar.gz if --tarball-uniqueid was
specified.
Added dumps of *_siginfo to the top of the txt files so you can
tell what signal was invoked.
Change-Id: Ib9ee6d83592d4b1bc90cb3419a05376a88d1ded9
when 'all' is specified in an allow or disallow section, it should erase
all values from the inverse section in the default config. E.G.
allow=all should erase any deny values from default config &
vice-versa
ASTERISK-27333 #close
Change-Id: I99219478fb98f08751d769daaee0b7795118a5a6
AMI goes from 3.2.0 to 4.0.0
ARI goes from 2.0.0 to 3.0.0
Copied UPGRADE.txt -> UPGRADE-15.txt
Created new UPGRADE.txt
Removed a log file that was accidentally checked in a while ago
Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7
The escalator works by creating a set of startup commands in cli.conf
that set up logger channels and issue the debug commands for the
subsystems specified. If asterisk is running when it is executed,
the same commands will be issued to the running instance. The original
cli.conf is saved before any changes are made and can be restored by
executing '$prog --reset'.
The log output will be stored in...
$astlogdir/message.$uniqueid
$astlogdir/debug.$uniqueid
$astlogdir/dtmf.$uniqueid
$astlogdir/fax.$uniqueid
$astlogdir/security.$uniqueid
$astlogdir/pjsip_history.$uniqueid
$astlogdir/sip_history.$uniqueid
Some minor tweaks were made to chan_sip, and res_pjsip_history
so their history output could be send to a log channel as packets
are captured.
A minor tweak was also made to manager so events are output to verbose
when "manager set debug on" is issued.
Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543
ast_loggrabber gathers log files from customizable search patterns,
optionally converts POSIX timestamps to a readable format and
tarballs the results.
Also a few tweaks were made to ast_coredumper.
Change-Id: I8bfe1468ada24c1344ce4abab7b002a59a659495
(cherry picked from commit c709152878)
This utility allows easy manipulation of asterisk coredumps.
* Configurable search paths and patterns for existing coredumps
* Can generate a consistent coredump from the running instance
* Can dump the lock_infos table from a coredump
* Dumps backtraces to separate files...
- thread apply 1 bt full -> <coredump>.thread1.txt
- thread apply all bt -> <coredump>.brief.txt
- thread apply all bt full -> <coredump>.full.txt
- lock_infos table -> <coredump>.locks.txt
* Can tarball corefiles and optionally delete them after processing
* Can tarball results files and optionally delete them after processing
* Converts ':' in coredump and results file names '-' to facilitate
uploading. Jira for instance, won't accept file names with colons
in them.
Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1].
[1] For *BSDs, the "devel/gdb" package might have to be installed to
get a recent gdb. The utility will check all instances of gdb
it finds in $PATH and if one isn't found that can run python, it
prints a friendly error.
Change-Id: I935d37ab9db85ef923f32b05579897f0893d33cd
(cherry picked from commit cb47b45560)
This change adds support for dual stack automatically. No
configuration is required and the IP address and version
in the SIP messages and SDP will be automatically changed
based on the transport over which the message is being
sent. RTP usage has also been changed to listen on both
IPv4 and IPv6 simultaneously to allow media to flow, and
to allow ICE support on both simultaneously. This also
allows failover between IPv6 and IPv4 to work as expected.
ASTERISK-26309 #close
Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d
Map the sip.conf general section legacy_useroption_parsing to the
new pjsip.conf global ignore_uri_user_options.
ASTERISK-26316
Reported by: Kevin Harwell
Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc
If you use the safe_asterisk script, it uses hardcoded defaults before
running configurable values from /etc/asterisk/startup.d. The hardcoded
default has TTY=9. Some containerized environments don't have such a
TTY, and safe_asterisk would stop.
The custom configuration from /etc/asterisk/startup.d/* isn't read until
after it stopped, so changing TTY in a custom config did not help.
This changeset changes safe_asterisk to continue if the TTY setting was
untouched and /dev/tty9 and /dev/vc/9 aren't found.
Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc
When using the migration script sip_to_pjsip.py, and your sip.conf is
configured with bindaddr=::, two transports are written to pjsip.conf, one for
0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4
and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface
like in chan_sip.
Furthermore, the script internal functions "build_host" and "split_hostport"
did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change
makes sure, even such addresses are parsed correctly.
ASTERISK-26309
Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48
When using the migration script sip_to_pjsip.py, cert_file was not migrated to
pjsip.conf. A previous change regarding this contained a copy/paste error.
ASTERISK-22374
Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b
A recent update had a copy/paste error where the unused variable 'val' was
being passed to the set_value function instead of the 'method' value itself.
This patch passes in the right variable.
ASTERISK-22374
Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06
When using the migration script sip_to_pjsip.py and tlsclientmethod is not set
in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to
overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is
offering/using not just TLSv1.0 but TLSv1.2 as well.
ASTERISK-22374
Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f
When using the migration script sip_to_pjsip.py, no section of type=system or
type=general were created. Therefore the keys compactheaders, timerb, timert1,
and useragent were not migrated to pjsip.conf.
ASTERISK-22374
Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1
When using the migration script sip_to_pjsip.py, session-timers=accept and
session-timers=refuse were mapped to wrong values.
ASTERISK-22374
Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092
When using the migration script sip_to_pjsip.py, now the (mandatory) username is
written to pjsip.conf, even if there was no (optional) authname in the register
string in sip.conf.
ASTERISK-22374
Change-Id: Ie53e1997104cd2674821688b8a8247249f5e156f
When using the migration script sip_to_pjsip.py and the register string
started with a transport in sip.conf - like tls://... - register was not parsed
correctly and therefore not migrated correctly to pjsip.conf.
ASTERISK-22374
Change-Id: I44c12104eea2bd8558ada6d25d77edfecd92edd2
When using the migration script sip_to_pjsip.py, those keys got missing. These
keys might appear several times and the function "merge_value" tried to collect
those. However, because these keys have different names in sip.conf and
pjsip.conf, "merge_value" was not able to find the new key name in sip.conf.
This change lets "merge_value" search with the old key name in sip.conf and
write with the new key name in pjsip.conf.
ASTERISK-22374
Change-Id: Ie53c5278ae6f1cb8fa7e96c5289877d46981d9d2
When using the migration script sip_to_pjsip.py, the externhost or externip of
sip.conf were erroneously written to Endpoints instead to Transports.
ASTERISK-22374
Change-Id: I2c5873386cfc388899fa9cf2368639dd12f1b8e4
When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and
minexpiry were not migrated to pjsip.conf.
ASTERISK-22374
Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b
When using the migration script sip_to_pjsip.py, encryption=yes got missing and
media_encryption=sdes was not written to pjsip.conf, because of a typo.
ASTERISK-22374
Change-Id: I0fc3e55dc512a57603ae0fef41baacccf2a35c05
When using the migration script sip_to_pjsip.py, both tos_sip and cos_sip got
missed, because of a typo. Therefore, cos and tos were not written to
pjsip.conf. Furthermore, that revealed a misuse of an internal function, caused
by a copy-and-paste error.
ASTERISK-22374
Change-Id: Id245ebadf70ab9776eb280c026288540af3af5c2
When using the migration script sip_to_pjsip.py, cert_file and ca_list_path were
not migrated to pjsip.conf.
ASTERISK-22374
Change-Id: I4612877d190b7f86a48698cefbf5c4db6c265825
The regular expression would match causing the code that handled
the line if it was merely a comment to never get executed.
Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819
When configuring SIP URIs in the pjsip.conf file it is
necessary to escape the semicolon so the parser does not
treat it as a comment. This change allows this to work in
the astconfigparser implementation.
A secondary bug where some data was lost if a configuration
option included a "=" in its value was also fixed.
A bug where sections would be considered equal despite
being different has also been fixed.
Change-Id: If229f656ef22050b50e7b34e90c4bffe796431f8
Since 5th November 2014, the master branch of libSRTP changed the prefix of
several member names and is not compatible with the source code in Asterisk
anymore. Therefore instead, this change checks out the latest version of the
libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as
backend. This makes AES-GCM and AES-IN possible.
ASTERISK-22131 #close
Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6
For all OSes:
* Disabled third-party codecs in pjproject and added
'--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the
configure options since we don't use the pjsip codec capability.
FreeBSD:
* Added FreeBSD support to install_prereq.
* Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make".
* Added __progname and environ to asterisk.exports.in.
* Reverted the use of ldconfig to create shared library symlinks to ln.
* Only enable epoll in pjproject if `uname -s` is Linux.
* Added a patch to pjproject to take the name of the 'make' command from
an environment variable if supplied. This is needed for the python bindings.
(merged by Teluu into pjproject trunk 5/3/2016)
FreeBSD support isn't complete. Still some general issues regarding
make/gmake having nothing to do with pjproject. With some handholding it DOES
build successfully.
CentOS:
Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH.
CentOS 6/7 32/64 build and run the pjsip testsuite successfully.
Ubuntu:
No changes required.
Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully.
Debian:
No changes required.
Debian 6/7/8 32/64 build and run the pjsip testsuite successfully.
There will utimately be a follow-up patch to create an install_prereq for
the testsuite as I've discovered a few missing requirements.
ASTERISK-25968 #close
Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c
check_installed_debs wasn't handling virtual packages like libsrtp-dev and
libresample-dev and on multiarch systems it was accidentally filtering out all
packages if any :i386 packages were found instead of just filtering out the
:i386 packages themselves.
Change-Id: Ifd68da0d1ee30cc84df14de3f9b9079d7c3cecda
RedHat/CentOS needs python-devel
Debian/Ubuntu needs automake, libsrtp-dev and python-dev
Ubuntu also needed libncurses5-dev for cmenuselect so while not
needed for pjproject, I adedd it anyway.
Change-Id: Idf5fa16e2d87c687439621507e122cb9461d7089
PJPROJECT has a function available to dump the compile time
options used when building the library.
* Add CLI "pjsip show buildopts" command.
* Update contrib/scripts/autosupport to get pjproject information.
Change-Id: Id93a6a916d765b2a2e5a1aeb54caaf83206be748
The spandspflow2pcap.py creates pcap files from fax.log files, generated
through 'fax set debug on' when receiving a fax. An example fax.log is
included as spandspflow2pcap.log.
The sipp-sendfax.xml SIPp scenario can be used to replay that fax with a
recent version of SIPp.
ASTERISK-25660 #close
Change-Id: I4de8f28b084055b482ab8a5b28d28b605b0ed526
When to install packages the indexed local is more old of the
version of software on the repository they have been upgraded by security
update then get the package will give 404 not found.
The patch prevent by update local index to repository for aptitude before
install.
ASTERISK-25495 #close
Reporte by: Rodrigo Ramírez Norambuena
Change-Id: I645959e553aac542805ced394cac2dca964051fa
This patch adds some minor tweaks for autosupport to update it for Asterisk 13.
This includes:
* Finally removing most references to Zaptel
* Adding support for some additional 'core' commands, and fixing nomenclature
that generally hasn't been used for some time
* Adding some PJSIP/SIP commands to gather endpoints/peers and active channels
Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1
(cherry picked from commit 9fc9777fa3)
To help in diagnosing mismatched modules and libraries, this
script scans for version, repository, and source information
and reports what is found.
ASTERISK-25376 #close
Reported by: Ashley Sanders
Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6
Asterisk needs the sqlite 3 library, which is package
sqlite-devel in CentOS. By adding this package to the
script, a problem with configure failing is resolved.
ASTERISK-25331 #close
Reported by: Kevin Harwell
Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec
If in Debian or system based, dont have aptitude installed the script do
nothing. This patch checked if aptitude installed, if not installed.
Also, if execute script with all packages installed yet, the script not show
nothing and return exit 1 because the command 'grep' get nothing from pipe from
'awk'.
ASTERISK-25113 #close
Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
Change-Id: Iebdff55805d3917166e5e08e0a1e2176f36ff27f
There are 3 ways that calls directly to standard allocator functions can
be dealt with:
1. Block their use, cause them to generate an error. This is the default.
2. Replace them with the Asterisk equivalent function calls.
3. Leave them alone.
This change allows one of these 3 options to be selected by any source.
The source just needs to define ASTMM_LIBC to ASTMM_BLOCK, ASTMM_REDIRECT,
or ASTMM_IGNORE to use option 1, 2 or 3 respectively. Normally ASTMM_BLOCK
is the correct option, so it is default when ASTMM_LIBC is not defined.
In some cases when building 3rd party code it is desirable to have it use
Asterisk functions, without changing the whole source - ASTMM_REDIRECT
accomplishes this. When using 3rd party libraries sometimes a static
inline function will make use of malloc or free. In these cases it may
be unsafe to replace the allocator in the header, as it's possible the
memory could be freed by the library using standard allocators. For
those cases ASTMM_IGNORE is needed.
Change-Id: I8afef4bc7f3b93914263ae27d3a5858b69663fc7
* The REF_DEBUG compiler flag no longer has any effect on code that uses
Astobj2. It is used to determine if reference debugging is enabled by
default. Reference debugging can be enabled or disabled in asterisk.conf.
* Caller information is provided in logger errors for ao2 bad magic numbers.
* Optimizes AO2 by merging internal functions with the public counterpart.
This was possible now that we no longer require a dual ABI.
ASTERISK-24974 #close
Reported by: Corey Farrell
Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1
On Debian based systems, the install_prereq tool uses a search command on
Debian that results in selecting both 64-bit and 32-bit packages. Besides the
waste of disk space, this can actually cause aptitude use 100% of memory on a
VM with 1GB of RAM as it tried to work out all of the 32-bit package
dependencies.
This patch filters out the 32-bit packages on a 64-bit machine, and leaves
32-bit machines alone.
ASTERISK-24048 #close
Reported by: Ben Klang
Tested by: Ben Klang, Matt Jordan
patches:
install_prereq_64-bit_compat.patch uploaded by Ben Klang (License 5876)
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General improvements to SIP to PJSIP conversion utility:
1) track default section of input file to allow parsing
an include file that doesn't specify a [section]
2) informatively handle case of assignment without [section]
3) correctly handle getting sections from included files
- [section]'s are inherited by included file
4) provide null string as default transport bind ip
5) gracefully handle missing portions of registration string
6) denote steps of operation during conversion and confirm
top level files as a convenience
ASTERISK-24474 #close
Review: https://reviewboard.asterisk.org/r/4280/
Reported by: John Kiniston
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This gets rid of most old libc free/malloc/realloc and replaces them
with ast_free and friends. When compiling with MALLOC_DEBUG you'll
notice it when you're mistakenly using one of the libc variants. For
the legacy cases you can define WRAP_LIBC_MALLOC before including
asterisk.h.
Even better would be if the errors were also enabled when compiling
without MALLOC_DEBUG, but that's a slightly more invasive header
file change.
Those compiling addons/format_mp3 will need to rerun
./contrib/scripts/get_mp3_source.sh.
ASTERISK-24348 #related
Review: https://reviewboard.asterisk.org/r/4015/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From now on, make install will overwrite safe_asterisk with the
latest version. You need to move any local modifications to files
inside /etc/asterisk/startup.d, if you have any.
See also commits r394939 and r397938.
ASTERISK-21965 #close
Patches:
safe_asterisk.patch uploaded by jkister (License 6232, modified by me)
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Cleans up the safe_asterisk script and adds the ASTSAFE_FOREGROUND
option that allows the debian asterisk init script to capture the
right pid.
* Drop the vim #modeline which wasn't used. Use test consistently
without the odd configure xno syntax. Double quote all paths.
General cleanup.
* Don't output message()s to the console but only to TTY if set.
* Allow TTY to be "no" as well as empty (debian compatibility with
debian/patches/safe_asterisk-config).
* Add option to export ASTSAFE_FOREGROUND=1 from the init script
that calls this to disable backgrounding. Debian uses a similar
method in debian/patches/safe_asterisk-nobg).
ASTERISK-23492 #close
Review: https://reviewboard.asterisk.org/r/3574/
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This patch does the following:
(1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
REF_DEBUG globally throughout Asterisk.
(2) The ref debug log file is now created in the AST_LOG_DIR directory.
Every run will now blow away the previous run (as large ref files
sometimes caused issues). We now also no longer open/close the file
on each write, instead relying on fflush to make sure data gets written
to the file (in case the ao2 call being performed is about to cause a
crash)
(3) It goes with a comma delineated format for the ref debug file. This
makes parsing much easier. This also now includes the thread ID of the
thread that caused ref change.
(4) A new python script instead for refcounting has been added in the
contrib/scripts folder.
(5) The old refcounter implementation in utils/ has been removed.
Review: https://reviewboard.asterisk.org/r/3377/
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The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter
and Tzafrir have pointed out numerous issues with the approach and have
propsed an alternative in r/2757. Since it's not a time critical issue and
is not worth holding up the release of 12 for it, I've gone ahead and reverted
r394939 from 12/trunk and re-opened ASTERISK-21965.
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Most, if not all, of the backing features of a conf file should now be
implemented (e.g. multi-line comments, includes, templates, etc...). A
few of the options still need to be mapped. Those are currently listed
in the 'sip_to_res_sip.py' file.
Things to do:
(1) There is more work to do here, at least for the sip.conf items that
aren't currently parsed. An issue will be created for that.
(2) All of the scripts should probably be passed through pylint and have
as many PEP8 issues fixed as possible.
(3) A public review is probably warranted at that point of the entire script.
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.
Review: https://reviewboard.asterisk.org/r/2708/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Alexander Traud
Jenkins2
Sean Bright
George Joseph
Joshua Colp
Tzafrir Cohen
Torrey Searle
Richard Mudgett
zuul
Walter Doekes
Kevin Harwell
George Joseph
Walter Doekes
Rodrigo Ramírez Norambuena
Matt Jordan
Scott Griepentrog
Rodrigo Ramírez Norambuena
Corey Farrell
Diederik de Groot
Scott Griepentrog
Sean Bright
Mark Michelson
David M. Lee
Kinsey Moore
Jason Parker