UpgradeNote: The safe_asterisk script now checks that, if it was run by the
root user, the /etc/asterisk/startup.d directory and all the files it contains
are owned by root. If the checks fail, safe_asterisk will exit with an error
and Asterisk will not be started. Additionally, the default logging
destination is now stderr instead of tty "9" which probably won't exist
in modern systems.
Resolves: #GHSA-v9q8-9j8m-5xwp
The suppress_moh_on_sendonly endpoint option should have been
defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES
in the alembic script instead of OPT_YESNO_T and YESNO_VALUES.
Also updated contrib/ast-db-manage/README.md to indicate that
AST_BOOL_VALUES should always be used and provided an example.
Resolves: #995
Normally, when one party in a call sends Asterisk an SDP with
a "sendonly" or "inactive" attribute it means "hold" and causes
Asterisk to start playing MOH back to the other party. This can be
problematic if it happens at certain times, such as in a 183
Progress message, because the MOH will replace any early media you
may be playing to the calling party. If you set this option
to "yes" on an endpoint and the endpoint receives an SDP
with "sendonly" or "inactive", Asterisk will NOT play MOH back to
the other party.
Resolves: #979
UserNote: The new "suppress_moh_on_sendonly" endpoint option
can be used to prevent playing MOH back to a caller if the remote
end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.
This patch introduces a new identifier for channels: tenantid. It's
a stringfield on the channel that can be used for general purposes. It
will be inherited by other channels the same way that linkedid is.
You can set tenantid in a few ways. The first is to set it in the
dialplan with the Set and CHANNEL functions:
exten => example,1,Set(CHANNEL(tenantid)=My tenant ID)
It can also be accessed via CHANNEL:
exten => example,2,NoOp(CHANNEL(tenantid))
Another method is to use the new tenantid option for pjsip endpoints in
pjsip.conf:
[my_endpoint]
type=endpoint
tenantid=My tenant ID
This is considered the best approach since you will be able to see the
tenant ID as early as the Newchannel event.
It can also be set using set_var in pjsip.conf on the endpoint like
setting other channel variable:
set_var=CHANNEL(tenantid)=My tenant ID
Note that set_var will not show tenant ID on the Newchannel event,
however.
Tenant ID has also been added to CDR. It's read-only and can be accessed
via CDR(tenantid). You can also get the tenant ID of the last channel
communicated with via CDR(peertenantid).
Tenant ID will also show up in CEL records if it has been set, and the
version number has been bumped accordingly.
Fixes: #740
UserNote: tenantid has been added to channels. It can be read in
dialplan via CHANNEL(tenantid), and it can be set using
Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
use the new tenantid option for pjsip endpoints (e.g., tenantid=My
tenant ID) so that it will show up in Newchannel events. You can set it
like any other channel variable using set_var in pjsip.conf as well, but
note that this will NOT show up in Newchannel events. Tenant ID is also
available in CDR and can be accessed with CDR(tenantid). The peer tenant
ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
as well if it has been set.
UpgradeNote: A new versioned struct (ast_channel_initializers) has been
added that gets passed to __ast_channel_alloc_ap. The new function
ast_channel_alloc_with_initializers should be used when creating
channels that require the use of this struct. Currently the only value
in the struct is for tenantid, but now more fields can be added to the
struct as necessary rather than the __ast_channel_alloc_ap function. A
new option (tenantid) has been added to endpoints in pjsip.conf as well.
CEL has had its version bumped to include tenant ID.
If the safe_asterisk script detects that the /var/lib/asterisk
directory doesn't exist, it now creates it with 755 permissions
instead of 770. safe_asterisk needing to create that directory
should be extremely rare though because it's normally created
by 'make install' which already sets the permissions to 755.
Resolves: #316
* Remove .gitreview and switch to pulling the main asterisk branch
version from configure.ac instead.
* Replace references to JIRA with GitHub.
* Other minor cleanup found along the way.
Resolves: #39
The following required columns were missing,
now added to the ps_endpoints table:
incoming_call_offer_pref
outgoing_call_offer_pref
stir_shaken_profile
ASTERISK-29453
Change-Id: I5cf565edf30195844d6acbc1e1de8c5f0d837568
This commit adds res_pjsip_geolocation which gives chan_pjsip
the ability to use the core geolocation capabilities.
This commit message is intentionally short because this isn't
a simple capability. See the documentation at
https://wiki.asterisk.org/wiki/display/AST/Geolocation
for more information.
THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON
USER FEEDBACK!
ASTERISK-30128
Change-Id: Ie2e2bcd87243c2cfabc43eb823d4427c7086f4d9
Rightly the use of wildcards in certificates is disallowed in accordance
with RFC5922. However, RFC2818 does make some allowances with regards to
their use when using subject alt names with DNS name types.
As such this patch creates a new setting for TLS transports called
'allow_wildcard_certs', which when it and 'verify_server' are both enabled
allows DNS name types, as well as the common name that start with '*.'
to match as a wildcard.
For instance: *.example.com
will match for: foo.example.com
Partial matching is not allowed, e.g. f*.example.com, foo.*.com, etc...
And the starting wildcard only matches for a single level.
For instance: *.example.com
will NOT match for: foo.bar.example.com
The new setting is disabled by default.
ASTERISK-30072 #close
Change-Id: If0be3fdab2e09c2a66bb54824fca406ebaac3da4
Correct typos of the following word families:
standard
increase
comments
valgrind
promiscuous
editing
libtonezone
storage
aggressive
whitespace
russellbryant
consecutive
peternixon
ASTERISK-29714
Change-Id: I9cafbf41b579c9c0c84c81719d2c4f900beec245
The search for a running asterisk when --running is used
has been greatly simplified and in the event it doesn't
work, you can now specify a pid to use on the command
line with --pid.
The search for asterisk modules when --tarball-coredumps
is used has been enhanced to have a better chance of finding
them and in the event it doesn't work, you can now specify
--libdir on the command line to indicate the library directory
where they were installed.
The DATEFORMAT variable was renamed to DATEOPTS and is now
passed to the 'date' utility rather than running DATEFORMAT
as a command.
The coredump and output files are now renamed with DATEOPTS.
This can be disabled by specifying --no-rename.
Several confusing and conflicting options were removed:
--append-coredumps
--conffile
--no-default-search
--tarball-uniqueid
The script was re-structured to make it easier for follow.
Change-Id: I674be64bdde3ef310b6a551d4911c3b600ffee59
In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
fallback use of the transport's bind address solve problems sending
media on systems that cannot send ipv4 packets on ipv6 sockets, and
certain other situations. This change extends both of these behaviors
to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
problems on these systems, introducing a new option
endpoint/t38_bind_udptl_to_media_address.
ASTERISK-29402
Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557
The behavior of max_contacts and remove_existing are connected. If
remove_existing is enabled, the soonest expiring contacts are removed.
This may occur when there is an unavailable contact. Similarly,
when remove_existing is not enabled, registrations from good
endpoints are rejected in favor of retaining unavailable contacts.
This commit adds a new AOR option remove_unavailable, and the effect
of this setting will depend on remove_existing. If remove_existing
is set to no, we will still remove unavailable contacts when they
exceed max_contacts, if there are any. If remove_existing is set to
yes, we will prioritize the removal of unavailable contacts before
those that are expiring soonest.
ASTERISK-29525
Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784
Adds the "auto" case which is valid with
both chan_sip dtmfmode and chan_pjsip's
dtmf_mode, adds subscribecontext to
subscribe_context conversion, and accounts
for cipher = ALL being invalid.
ASTERISK-29459
Change-Id: Ie27d6606efad3591038000e5f3c34fa94730f6f2
With newer version of linux /var/run/ is a symlink to /run/ that has
been turned into tmpfs.
Added note that if asterisk has to bind to a specific IP that
systemd has to wait until the network is up.
Added note on how to make sure that the environment variable
HOSTNAME is included.
ASTERISK-29216
Reported by: Mark Petersen
Tested by: Mark Petersen
Change-Id: Ib3e560655befd3e99eec743687144f5569533379
* Wildcards in #includes are now properly expanded
* Implement operators for Section class to allow sorting
ASTERISK-29142 #close
Change-Id: I9b9cd95f4cbe5c24506b75d17173c5aa1a83e5df
Ubuntu 20.10 does not come with GMime 2.6. Ubuntu 16.04 LTS does not
come with GMime 3.0. aptitude ignores any missing package. Therefore,
it installs the correct package(s). However, in Ubuntu 18.04 LTS and
Ubuntu 20.04 LTS, both versions are installed alongside although only
one is really needed.
Change-Id: Ic58aa9f2e131d94671f286f17dbd61e1ccbabcb7
Implemention of History-Info capable of interworking with Diversion
Header following RFC7544
ASTERISK-29027 #close
Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6
Currently, the ps_contacts table's reg_server column in realtime database type is varchar(20).
This is fine for normal cases, but if the hostname is longer than 20, it returns error and then
failed to register the contact address of the peer.
Normally, 20 characters limitation for the hostname is fine, but with the cloud env.
So, increased the size to 255.
ASTERISK-29056
Change-Id: Iac52c8c35030303cfa551bb39f410b33bffc507d
If you run ast_coredumper --tarball-coredumps in the same directory
as the actual coredump, tar can fail because the link to the
actual coredump becomes recursive. The resulting tarball will
have everything _except_ the coredump (which is usually what
you need)
There's also an issue that the directory name in the tarball
is the same as the coredump so if you extract the tarball the
directory it creates will overwrite the coredump.
So:
* Made the link to the coredump use the absolute path to the
file instead of a relative one. This prevents the recursive
link and allows tar to add the coredump.
* The tarballed directory is now named <coredump>.output instead
of just <coredump> so if you expand the tarball it won't
overwrite the coredump.
Change-Id: I8b3eeb26e09a577c702ff966924bb0a2f9a759ea
With the latest Linux, 'ifconfig' is not installed on default anymore.
Furthermore, the output of the current net-tools 'ifconfig' changed.
Therefore, parsing failed. This update uses 'ip addr show' instead.
Finally, the service for the external IP changed.
Change-Id: I9b1a7c3f457e3553b50a3e9a55524e40d70245a0
This change renames the codec preference endpoint options.
incoming_offer_codec_prefs becomes codec_prefs_incoming_offer
to keep the options together when showing an endpoint.
Change-Id: I6202965b4723777f22a83afcbbafcdafb1d11c8d
Added a new configuration option for PJSIP endpoints - stir_shaken. If
set to yes, then STIR/SHAKEN support will be added to inbound and
outbound INVITEs. The default is no. Alembic has been updated to include
this option.
Previously the dialplan function was not trimming the whitespace from
the parameters it recieved. Now it does.
Also added a conditional that, when TEST_FRAMEWORK is enabled, the
timestamp in the identity header will be overlooked. This is just for
testing, since the testsuite will rely on a SIPp scenario with a preset
identity header to trigger the MISMATCH result.
Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1
This commit adds the endpoint options required to control
Advanced Codec Negotiation.
incoming_offer_codec_prefs
outgoing_offer_codec_prefs
incoming_answer_codec_prefs
outgoing_answer_codec_prefs
The documentation may need tweaking and some additional edits
added, especially for the "answer" prefs. That'll be handled
when things finalize.
This commit is safe to merge as it doens't alter any existing
functionality nor does it alter the previous codec negotiation
work which may now be obsolete.
Change-Id: I920ba925d7dd36430dfd2ebd9d82d23f123d0e11
Currently when the pjsip making an outgoing request, it keep adding the
rport parameter in a request message as a default.
This causes unexpected rport handle at the other end.
Added option for disable this behaviour in the pjsip.conf.
This is a system option, but working as a gloabl option.
ASTERISK-28959
Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc
The Python/C API is used only if the Test Framework was enabled in Asterisk
'make menuselect'. The Test Framework is available only if the Developer Mode
was enabled in Asterisk './configure --enable-dev-mode'. And that Python/C API
is used only if the PJProject was found and not disabled in Asterisk; the user
did not go for './configure --without-pjproject'.
Furthermore, because version 2 of that Python/C API is required (currently) and
because some platforms do not offer a generic version 2, the script searches
for 2.7 explicitly as well.
To avoid version mismatch between the Python/C API and the Python environment,
the script searches for the latter in the same versions, in the same the order
as well. Because this Python/C API is just for (some) Asterisk contributors,
the script also goes for the Python 3 environment as a last resort for all
other Asterisk users. This allows 'make full' even on minimal installations of
Ubuntu 18.04 LTS and newer.
Because the Python/C API is Asterisk contributor specific, the Python packages
are removed from the script './contrib/scripts/install_prereq' as this script
is intended for Asterisk users. Asterisk contributors have to install much more
packages in any case, like:
sudo apt install autoconf automake git git-review python2.7-dev
ASTERISK-28824
ASTERISK-27717
Change-Id: Id46d357e18869f64dcc217b8fdba821b63eeb876
This patch makes it so ast_coredumper now outputs the following information to
a *-info.txt file when processing a core file:
asterisk version and "built by" string
BUILD_OPTS
system start, and last reloaded date/time
taskprocessor list
equivalent of "bridge show all"
equivalent of "core show channels verbose"
Also a slight modification was made when trying to obtain the pid(s) of a
running Asterisk. If it fails to retrieve any it now reports an error.
Change-Id: I54f35c19ab69b8f8dc78cc933c3fb7c99cef346b
Currently aptitude is installed using interactive mode. This patch
changes this to use the non-interactive mode as it can block
automatic dependencies installation, ex: CI, Docker build.
ASTERISK-28726 #close
Change-Id: I271ee00d230513a6f044810351a32d83b2181133
Fix use of frame-level wildcard usage in suppression file.
ASTERISK-27243 #close
Reported-by: Richard Kenner
Change-Id: I1c0c64c5f305d2c9aa124e11f1f64a2eec52dc51
Allow the list of files to be played to be provided explicitly in the
music class's configuration. The primary driver for this change is to
allow URLs to be used for MoH.
Change-Id: I9f43b80b43880980b18b2bee26ec09429d0b92fa
Added a new PJSIP global setting called norefersub.
Default is true to keep support working as before.
res_pjsip_refer: Configures PJSIP norefersub capability accordingly.
Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.
This is useful for Cisco switches that do not follow RFC4488.
ASTERISK-28375 #close
Reported-by: Dan Cropp
Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9
When trying to upgrade using alembic, a couple different errors kept
popping up that prevented the upgrade. An additional parameter was
needed when changing the schema for mwi_subscribe_replaces_unsolicited
from an integer to an enum. When changing from a string to an enum, the
type needed to be cast for postgresql. The other issue was a parameter
being used during column creation that did not exist.
After fixing the upgrade process, it revealed errors with the downgrade
process. One was a variable not being defined in the downgrade function,
and the other was tables not existing when using MySQL. This was due to
a context check that should have encompassed MySQL, but in the end was
not doing so.
Change-Id: Ib4d70cf3ce5080023a50be496272a777b55d6c8e
When variable ALTCONF is defined, the command start prints the message
"Unable to open specified master config file '"/etc/asterisk/asteris..."
and use default configurations.
ASTERISK-28332
Change-Id: I7595e582a0ee2c1051ea35435e247e27906957ef
chan_sip will always ignore 183 responses that do not contain SDP
however, chan_pjsip will currently always translate it into a
183 with SDP. This new flag allows chan_pjsip to have the same
behavior as chan_sip.
ASTERISK-28322 #close
Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a
