This patch adds a new features to the endpoint identifier module,
'match_header'. When set, inbound requests are matched by a provided SIP
header: value pair. This option works in conjunction with the existing
'match' configuration option, such that if any 'match*' attribute
matches an inbound request, the request is associated with the specified
endpoint.
Since this module now identifies by more than just IP address,
appropriate renaming of the module and/or variables can be done in a
non-release branch.
ASTERISK-26863 #close
Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453
(cherry picked from commit 30f52d79d7)
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 implements SRV support for the IP based endpoint
identifier module. All possible addresses through SRV are looked
up and added as matches. If no SRV records are available a
fallback to normal host resolution is done. If an IP address
is provided then no SRV lookup occurs.
This is configured using the "srv_lookups" option on the
identify section and defaults to "yes".
ASTERISK-26693
Change-Id: I6b641e275bf96629320efa8b479737062aed82ac
Pyflake is a python (2) source checker. This patch fixes various
(mostly trivial) errors and warnings it reports.
Change-Id: Ia35c5ac61751b927814cf693994c632c412386ea
When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.
The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.
The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.
ASTERISK-26423 #close
Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc
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
cdr, config and voicemail are all separate alembic trees. Because
alembic's default is to use a table named 'alembic_version' to store
the current tree revision, the 3 trees can't exist in the same schema
without stepping on each other.
Now each tree uses 'alembic_version_<tree_name>' as the version table.
Each tree's env.py script now first checks for 'alembic_version'. If
it finds it AND its revision is in the tree's history, the script
renames it to 'alembic_version_<tree_name>'. Regardless, the script
then continues with the migration using 'alembic_version_<tree_name>'
and creates that table if it's not found. The result is that if an
existing 'alembic_version' table was found but it didn't belong to this
tree, it's left alone and 'alembic_version_<tree_name>' is used or
created.
WARNING: If multiple trees are using the same schema, they MUST NOT
CRU or D any objects with names that might exist in the other trees.
An example would be 'yesno_values' type. If two trees perform
operations on it, one tree could pull it out from under the other.
Thankfully we currently don't share any names among cdr, config and
voicemail.
NOTE: Since the env.py scripts in each tree were identical, a common
env.py has been placed in the ast-db-manage directory and a symlink
to it has been placed in each tree directory.
ASTERISK-24311 #close
Reported-by: Dafi Ni
Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898
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
This implements the chan_sip legacy_useroption_parsing option but with a
better name.
* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.
ASTERISK-26316 #close
Reported by: Kevin Harwell
Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
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
This patch add config to pjsip by endpoint.
;preferred_codec_only=yes
; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.
ASTERISK-26317 #close
Reported by: AaronAn
Tested by: AaronAn
Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762
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
