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
In Asterisk configuration, a multiline comment starts with ;-- as long as it is
not followed by another dash (i.e. ;--- is not a multiline comment).
ASTERISK-28323 #close
Change-Id: I32dc38e0fac01d3c0805d27d35d2365a7c37ca72
To prevent one subsystem's taskprocessors from causing others
to stall, new capabilities have been added to taskprocessors.
* Any taskprocessor name that has a '/' will have the part
before the '/' saved as its "subsystem".
Examples:
"sorcery/acl-0000006a" and "sorcery/aor-00000019"
will be grouped to subsystem "sorcery".
"pjsip/distributor-00000025" and "pjsip/distributor-00000026"
will bn grouped to subsystem "pjsip".
Taskprocessors with no '/' have an empty subsystem.
* When a taskprocessor enters high-water alert status and it
has a non-empty subsystem, the subsystem alert count will
be incremented.
* When a taskprocessor leaves high-water alert status and it
has a non-empty subsystem, the subsystem alert count will be
decremented.
* A new api ast_taskprocessor_get_subsystem_alert() has been
added that returns the number of taskprocessors in alert for
the subsystem.
* A new CLI command "core show taskprocessor alerted subsystems"
has been added.
* A new unit test was addded.
REMINDER: The taskprocessor code itself doesn't take any action
based on high-water alerts or overloading. It's up to taskprocessor
users to check and take action themselves. Currently only the pjsip
distributor does this.
* A new pjsip/global option "taskprocessor_overload_trigger"
has been added that allows the user to select the trigger
mechanism the distributor uses to pause accepting new requests.
"none": Don't pause on any overload condition.
"global": Pause on ANY taskprocessor overload (the default and
current behavior)
"pjsip_only": Pause only on pjsip taskprocessor overloads.
* The core pjsip pool was renamed from "SIP" to "pjsip" so it can
be properly grouped into the "pjsip" subsystem.
* stasis taskprocessor names were changed to "stasis" as the
subsystem.
* Sorcery core taskprocessor names were changed to "sorcery" to
match the object taskprocessors.
Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56
The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out
the ContactStatus AMI event when a contact is updated.
Thist change broke things which rely on old behavior.
This patch adds a new PJSIP global configuration option
'send_contact_status_on_update_registration' to be able to preserve old
ContactStatus behavior.
By default new behavior, i.e. the ContactStatus event will not be sent when a
device refreshes its registration.
Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46
In order to get a dump of the running process, we need to find the
pid of the main asterisk process. This can be tricky if there are
also instances of "asterisk -r" running or if an alternate location
for asterisk.conf was specified on the command line with the -C
option that also specified an alternation location for the pid file.
So now...
1. We find the asterisk executable with "which" or the --asterisk-bin
command line option.
2. If there's only 1 process with an executable path that matches,
we use that pid. If not...
3. We try "<asterisk-bin> -rx 'core show settings'" and parse the
output to find the pidfile, then read that for the pid. If that
didn't work...
4. We get a list of all the pids matching <asterisk-bin> and look
in /proc/<pid>/cmdline for a -C argument and retry the "core show
settings" using the same -C option. We can't parse the output
of "ps" to get the -C path because it may contain spaces. The
contents of /proc/<pid>/cmdline are delimited by NULLs. For BSDs
we may have to mount /proc first. :(
ASTERISK-28221
Reported by: Andrew Nagy
Change-Id: I8aa1f3f912f949df2b5348908803c636bde1d57c
Using the --quiet or -q option in conjonction with /dev/stdout as the output
file allow the output to be used as a valid configuration.
Given a script that generates a valid sip.conf I can pipe the output of that
script into `sip_to_pjsip.py -q /dev/stdin /dev/stdout`. This allow me to use
that piped command in my pjsip.conf using the `exec` command.
ASTERISK-28136
Change-Id: I7b0e2e90e2549f3f8e01dc96701f111b5874c88d
This patch adds new options 'trust_connected_line' and 'send_connected_line'
to the endpoint.
The option 'trust_connected_line' is to control if connected line updates
are accepted from this endpoint.
The option 'send_connected_line' is to control if connected line updates
can be sent to this endpoint.
The default value is 'yes' for both options.
Change-Id: I16af967815efd904597ec2f033337e4333d097cd
Given a sip.conf with the following content:
setvar FOO=1
setvar BAR=42
I want my generated pjsip.conf to containt the following set_vars
set_var FOO=1
set_var BAR=42
in the matching endpoint section.
Change-Id: I6c822401fda4133c3b44bf31e655b4eb939d4d26
Add a new global flag to res_pjsip to allow the callerid to be used
as the username in the contact header. This allows chan_pjsip to have
the same behavour as chan_sip
ASTERISK-28087 #close
Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95
This change implements a few different generic things which were brought
on by Google Voice SIP.
1. The concept of flow transports have been introduced. These are
configurable transports in pjsip.conf which can be used to reference a
flow of signaling to a target. These have runtime configuration that can
be changed by the signaling itself (such as Service-Routes and
P-Preferred-Identity). When used these guarantee an individual connection
(in the case of TCP or TLS) even if multiple flow transports exist to the
same target.
2. Service-Routes (RFC 3608) support has been added to the outbound
registration module which when received will be stored on the flow
transport and used for requests referencing it.
3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been
added to the outbound registration module. If a P-Associated-URI header
is received it will be used on requests as the P-Preferred-Identity.
4. Configurable outbound extension support has been added to the outbound
registration module. When set the extension will be placed in the
Supported header.
5. Header parameters can now be configured on an outbound registration
which will be placed in the Contact header.
6. Google specific OAuth / Bearer token authentication
(draft-ietf-sipcore-sip-authn-02) has been added to the outbound
registration module.
All functionality changes are controlled by pjsip.conf configuration
options and do not affect non-configured pjsip endpoints otherwise.
ASTERISK-27971 #close
Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
This allows us to process AO2 statistics for total objects, memory
usage, memory overhead and lock usage.
* Install refstats.py and reflocks.py into the Asterisk scripts folder.
* Enable support for reflocks.py without DEBUG_THREADS.
Steal a bit from the ao2 magic to flag when an object lock is used.
Remove 'lockobj' from reflocks.py since we can now record 'used' or
'unused' for those objects.
Add comments to explain thread safety of the 'struct __priv_data'
bitfields.
Change-Id: I84e9d679cc86d772cc97c888d9d856a17e0d3a4a
When DEBUG_THREADS is enabled we can know if the astobj2 mutex / rwlock
was ever used, so it can be recorded in the REF_DEBUG destructor entry.
Create contrib/scripts/reflocks.py to process locking used by
allocator. This can be used to identify places where
AO2_ALLOC_OPT_LOCK_NOLOCK should be used to reduce memory usage.
Change-Id: I2e3cd23336a97df2692b545f548fd79b14b53bf4
In the original commit introducing the feature the column in the alembic
script was called 'suppress_q850_reason_header'.
In the code however the option is called 'suppress_q850_reason_headers'
(trailing 's'). This leads to errors when ARI push configuration is used.
Change-Id: Ie84808adbca6fcc9136556e4f5d741adbef5d14f
This is removed in favor of ./configure --with-jansson-bundled. The
install-unpackaged command would only install jansson once, so once
installed it would never update, where the bundled copy will be kept up
to date.
Change-Id: Ideab1f65419608d3795aa608e9da873823cc42d3
When mobile SIP clients register with Asterisk that use some sort of
push notifications, the URI can get quite lengthy due to the
additional push-service annotations (things like tokens, pn-type, etc.)
contained in it.
ASTERISK-28022 #close
Change-Id: I4c7ceadc3bb405f3daf722641c8cd5ca4188cc37
A new option 'suppress_q850_reason_headers' has been added to the
endpoint object. Some devices can't accept multiple Reason headers and
get confused when both 'SIP' and 'Q.850' Reason headers are received.
This option allows the 'Q.850' Reason header to be suppressed.
The default value is 'no'.
ASTERISK-27949
Reported-by: Ross Beer
Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1
Torrey Searle
Sungtae Kim
George Joseph
Alexander Traud
Ben Ford
Kevin Harwell
Sean Bright
Sylvain Afchain
Joshua Colp
Snuffy
Pascal Cadotte Michaud
Rodrigo Ramírez Norambuena
Walter Doekes
Dan Cropp
cirillor
Torrey Searle
Alexei Gradinari
Richard Mudgett
Nick French
Corey Farrell
Florian Floimair