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497 Commits

Author SHA1 Message Date
Andreas Eversberg 688f7eff12 undo icsi contact header support
use this in registration block instead:

contact_header_params=+g.3gpp.icsi-ref="urn\%3Aurn-7\%3A3gpp-service.ims.icsi.mmtel"\;+g.3gpp.smsip\;audio\;+sip.instance="<urn:gsma:imei:35876110-027790-0>"
2024-05-27 12:11:44 +02:00
Eric Wild cb8209390a WIP icsi conctact header support 2024-05-22 09:25:59 +02:00
Andreas Eversberg d845adafeb Fixup: Volte support
Apply correct transport to IPSec (MT calls)

Note:
MT calls work, but Kamailio still uses non-expected transport.
2024-05-21 18:43:15 +02:00
Andreas Eversberg 6d28717a59 fixup: Volte support
P-Access-Network-Info header only when registering via IPSEC.
2024-05-21 14:26:31 +02:00
Andreas Eversberg 68208848a0 fixup: VoLTE support for outgoing and incoming calls
Fixes crash, if ims_aka is not enabled at endpoint.
2024-05-17 09:40:13 +02:00
Andreas Eversberg df9f6c6e32 fixup: VoLTE support for outgoing and incoming calls
Add P-Access-Network-Info header.
2024-05-16 15:12:21 +02:00
Andreas Eversberg 379bac0ecb fixup: Volte support
Add P-Access-Network-Info header.
2024-05-16 15:10:45 +02:00
Andreas Eversberg 7268e8b646 fixup: AMI test client for VoLTE development 2024-05-16 15:09:16 +02:00
Andreas Eversberg b0e65f0cb2 WIP: VoLTE support for outgoing and incoming calls 2024-05-16 11:22:55 +02:00
Eric Wild 450b38031a WIP add amr support
Based on
http://ftp.debian.org/debian/pool/main/a/asterisk/asterisk_20.6.0~dfsg+~cs6.13.40431414.orig-Xamr.tar.xz

requires
            libopencore-amrnb-dev \
            libopencore-amrwb-dev \
            libvo-amrwbenc-dev

license "unlicense" according to source

This is free and unencumbered software released into the public domain.

Anyone is free to copy, modify, publish, use, compile, sell, or
distribute this software, either in source code form or as a compiled
binary, for any purpose, commercial or non-commercial, and by any
means.

In jurisdictions that recognize copyright laws, the author or authors
of this software dedicate any and all copyright interest in the
software to the public domain. We make this dedication for the benefit
of the public at large and to the detriment of our heirs and
successors. We intend this dedication to be an overt act of
relinquishment in perpetuity of all present and future rights to this
software under copyright law.

THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
IN NO EVENT SHALL THE AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR
OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE,
ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
OTHER DEALINGS IN THE SOFTWARE.

For more information, please refer to <http://unlicense.org/>
2024-05-16 09:31:10 +02:00
Andreas Eversberg 4dc9a6ffc2 fixup: Volte support
Fix crash when cleaning up transport. (re-register)
2024-05-16 09:26:37 +02:00
Andreas Eversberg 717b64fb17 fixup: Volte support
Fixes uninitialized memory, found by valgrind.
2024-05-15 17:01:30 +02:00
Andreas Eversberg 1eb1d32f40 fixup: Volte support 2024-05-14 13:41:19 +02:00
Andreas Eversberg cedd34759a AMI test client for VoLTE development 2024-05-10 15:51:58 +02:00
Andreas Eversberg 4cd2c70ee9 WIP: Volte support for outgoing SIP registration 2024-05-10 15:51:58 +02:00
Andreas Eversberg ff3c350049 Hack to make registration without prior unregistration
It needs to be investigated why this is done an why it does not work.
2024-05-10 15:51:58 +02:00
Andreas Eversberg 093edc657c Add option to disable automatic registration
Use "manual_register=yes" in registration block of pjsip.conf to disable
automatic registration. Then registration must be triggered by AMI
interface.
2024-05-10 15:51:58 +02:00
Andreas Eversberg d23409eecd Add support for IMS AKA authentication configuration 2024-05-10 15:51:58 +02:00
Andreas Eversberg 2e2861402a For Work in Progress: Unstage some files 2024-05-10 15:51:58 +02:00
Andreas Eversberg fd747c4dbb WIP: Vocal EVS Codec integration
Related: SY#6825
2024-05-10 15:51:58 +02:00
Asterisk Development Team af5acc405d Update for 20.7.0 2024-03-18 13:21:48 +00:00
Asterisk Development Team 9b6c74e82a Update for 20.7.0-rc2 2024-03-11 17:17:22 +00:00
George Joseph d7597e5148 res_pjsip_stir_shaken.c: Add checks for missing parameters
* Added checks for missing session, session->channel and rdata
  in stir_shaken_incoming_request.

* Added checks for missing session, session->channel and tdata
  in stir_shaken_outgoing_request.

Resolves: #645
2024-03-11 11:07:00 -06:00
Asterisk Development Team 959babc472 Update for 20.7.0-rc1 2024-03-07 14:17:30 +00:00
Naveen Albert 40fd161575 app_dial: Add dial time for progress/ringing.
Add a timeout option to control the amount of time
to wait if no early media is received before giving
up. This allows aborting early if the destination
is not being responsive.

Resolves: #588

UserNote: The timeout argument to Dial now allows
specifying the maximum amount of time to dial if
early media is not received.

(cherry picked from commit e4adc962ca)
2024-03-07 14:17:23 +00:00
Naveen Albert 5ade71ec9a app_voicemail: Properly reinitialize config after unit tests.
Most app_voicemail unit tests were not properly cleaning up
after themselves after running. This led to test mailboxes
lingering around in the system. It also meant that if any
unit tests in app_voicemail that create mailboxes were executed
and the module was not unloaded/loaded again prior to running
the test_voicemail_vm_info unit test, Asterisk would segfault
due to an attempt to copy a NULL string.

The load_config test did actually have logic to reinitialize
the config after the test. However, this did not work in practice
since load_config() would not reload the config since voicemail.conf
had not changed during the test; thus, additional logic has been
added to ensure that voicemail.conf is truly reloaded, after any
unit tests which modify the users list.

This prevents the SEGV due to invalid mailboxes lingering around,
and also ensures that the system state is restored to what it was
prior to the tests running.

Resolves: #629
(cherry picked from commit 082966bdad)
2024-03-07 14:17:23 +00:00
Shaaah 395bcab787 app_queue.c : fix "queue add member" usage string
Fixing bracket placement in the "queue add member" cli usage string.

(cherry picked from commit c6ff7a6a51)
2024-03-07 14:17:23 +00:00
Naveen Albert 3469e515de app_voicemail: Allow preventing mark messages as urgent.
This adds an option to allow preventing callers from leaving
messages marked as 'urgent'.

Resolves: #619

UserNote: The leaveurgent mailbox option can now be used to
control whether callers may leave messages marked as 'Urgent'.

(cherry picked from commit 3c42b141d7)
2024-03-07 14:17:23 +00:00
Sean Bright f8da5a2ba3 res_pjsip: Use consistent type for boolean columns.
This migrates the relevant schema objects from the `('yes', 'no')`
definition to the `('0', '1', 'off', 'on', 'false', 'true', 'yes', 'no')`
one.

Fixes #617

(cherry picked from commit 93e40968c3)
2024-03-07 14:17:23 +00:00
George Joseph f02ac33364 .github: Remove timeout-minutes from gatetests
(cherry picked from commit 9f182e9f58)
2024-03-07 14:17:23 +00:00
George Joseph 44563b5043 attestation_config.c: Use ast_free instead of ast_std_free
In as_check_common_config, we were calling ast_std_free on
raw_key but raw_key was allocated with ast_malloc so it
should be freed with ast_free.

Resolves: #636
(cherry picked from commit 267348bee3)
2024-03-07 14:17:23 +00:00
George Joseph efc2124594 Makefile: Add stir_shaken/cache to directories created on install
The default location for the stir_shaken cache is
/var/lib/asterisk/keys/stir_shaken/cache but we were only creating
/var/lib/asterisk/keys/stir_shaken on istall.  We now create
the cache sub-directory.

Resolves: #634
(cherry picked from commit df0221b53d)
2024-03-07 14:17:23 +00:00
George Joseph be5fd6180c .github: Pass only single GATETEST_COMMAND to AsteriskGateComposite
(cherry picked from commit eed8288d0c)
2024-03-07 14:17:23 +00:00
George Joseph fd27df9479 Stir/Shaken Refactor
Why do we need a refactor?

The original stir/shaken implementation was started over 3 years ago
when little was understood about practical implementation.  The
result was an implementation that wouldn't actually interoperate
with any other stir-shaken implementations.

There were also a number of stir-shaken features and RFC
requirements that were never implemented such as TNAuthList
certificate validation, sending Reason headers in SIP responses
when verification failed but we wished to continue the call, and
the ability to send Media Key(mky) grants in the Identity header
when the call involved DTLS.

Finally, there were some performance concerns around outgoing
calls and selection of the correct certificate and private key.
The configuration was keyed by an arbitrary name which meant that
for every outgoing call, we had to scan the entire list of
configured TNs to find the correct cert to use.  With only a few
TNs configured, this wasn't an issue but if you have a thousand,
it could be.

What's changed?

* Configuration objects have been refactored to be clearer about
  their uses and to fix issues.
    * The "general" object was renamed to "verification" since it
      contains parameters specific to the incoming verification
      process.  It also never handled ca_path and crl_path
      correctly.
    * A new "attestation" object was added that controls the
      outgoing attestation process.  It sets default certificates,
      keys, etc.
    * The "certificate" object was renamed to "tn" and had it's key
      change to telephone number since outgoing call attestation
      needs to look up certificates by telephone number.
    * The "profile" object had more parameters added to it that can
      override default parameters specified in the "attestation"
      and "verification" objects.
    * The "store" object was removed altogther as it was never
      implemented.

* We now use libjwt to create outgoing Identity headers and to
  parse and validate signatures on incoming Identiy headers.  Our
  previous custom implementation was much of the source of the
  interoperability issues.

* General code cleanup and refactor.
    * Moved things to better places.
    * Separated some of the complex functions to smaller ones.
    * Using context objects rather than passing tons of parameters
      in function calls.
    * Removed some complexity and unneeded encapsuation from the
      config objects.

Resolves: #351
Resolves: #46

UserNote: Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.

UpgradeNote: The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed.  The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information.  This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added.  Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.

(cherry picked from commit e6c7f1aee0)
2024-03-07 14:17:23 +00:00
Sean Bright 507c28f268 alembic: Synchronize alembic heads between supported branches.
This adds a dummy migration to 18 and 20 so that our alembic heads are
synchronized across all supported branches.

In this case the migration we are stubbing (24c12d8e9014) is:

775352ee6c
(cherry picked from commit d122225279)
2024-03-07 14:17:23 +00:00
Sebastian Jennen 77808edbd4 translate.c: implement new direct comp table mode
The new mode lists for each codec translation the actual real cost in cpu microseconds per second translated audio.
This allows to compare the real cpu usage of translations and helps in evaluation of codec implementation changes regarding performance (regression testing).

- add new table mode
- hide the 999999 comp values, as these only indicate an issue with transcoding
- hide the 0 values, as these also do not contain any information (only indicate a multistep transcoding)

Resolves: #601
(cherry picked from commit a47acba99b)
2024-03-07 14:17:23 +00:00
Shyju Kanaprath 2868366f00 README.md: Removed outdated link
Removed outdated link http://www.quicknet.net from README.md

cherry-pick-to: 18
cherry-pick-to: 20
cherry-pick-to: 21
(cherry picked from commit 667b5ee192)
2024-03-07 14:17:23 +00:00
Sean Bright 02a17926c4 strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
If a dynamic string is created with an initial length of 0,
`ast_str_buffer(…)` will return an invalid pointer.

This was a secondary discovery when fixing #65.

(cherry picked from commit 31fc2877cb)
2024-03-07 14:17:23 +00:00
George Joseph 4152f67a0b .github: Add force_cherry_pick option to Releaser
(cherry picked from commit bbafb63bb5)
2024-03-07 14:17:23 +00:00
George Joseph 41ae2d754f .github: Remove start_version from Releaser
(cherry picked from commit 7bd9f2b5a5)
2024-03-07 14:17:23 +00:00
romryz a494b61fe2 res_rtp_asterisk.c: Correct coefficient in MOS calculation.
Media Experience Score relies on incorrect pseudo_mos variable
calculation. According to forming an opinion section of the
documentation, calculation relies on ITU-T G.107 standard:

    https://docs.asterisk.org/Deployment/Media-Experience-Score/#forming-an-opinion

ITU-T G.107 Annex B suggests to calculate MOS with a coefficient
"seven times ten to the power of negative six", 7 * 10^(-6). which
would mean 6 digits after the decimal point. Current implementation
has 7 digits after the decimal point, which downrates the calls.

Fixes: #597
(cherry picked from commit 335b925fac)
2024-03-07 14:17:23 +00:00
Naveen Albert 70ff49c474 dsp.c: Fix and improve potentially inaccurate log message.
If ast_dsp_process is called with a codec besides slin, ulaw,
or alaw, a warning is logged that in-band DTMF is not supported,
but this message is not always appropriate or correct, because
ast_dsp_process is much more generic than just DTMF detection.

This logs a more generic message in those cases, and also improves
codec-mismatch logging throughout dsp.c by ensuring incompatible
codecs are printed out.

Resolves: #595
(cherry picked from commit 59df4892ad)
2024-03-07 14:17:23 +00:00
George Joseph f770c9c92c pjsip show channelstats: Prevent possible segfault when faxing
Under rare circumstances, it's possible for the original audio
session in the active_media_state default_session to be corrupted
instead of removed when switching to the t38/image media session
during fax negotiation.  This can cause a segfault when a "pjsip
show channelstats" attempts to print that audio media session's
rtp statistics.  In these cases, the active_media_state
topology is correctly showing only a single t38/image stream
so we now check that there's an audio stream in the topology
before attempting to use the audio media session to get the rtp
statistics.

Resolves: #592
(cherry picked from commit cb057a6381)
2024-03-07 14:17:23 +00:00
George Joseph 6df5fbee65 Reduce startup/shutdown verbose logging
When started with a verbose level of 3, asterisk can emit over 1500
verbose message that serve no real purpose other than to fill up
logs. When asterisk shuts down, it emits another 1100 that are of
even less use. Since the testsuite runs asterisk with a verbose
level of 3, and asterisk starts and stops for every one of the 700+
tests, the number of log messages is staggering.  Besides taking up
resources, it also makes it hard to debug failing tests.

This commit changes the log level for those verbose messages to 5
instead of 3 which reduces the number of log messages to only a
handful. Of course, NOTICE, WARNING and ERROR message are
unaffected.

There's also one other minor change...
ast_context_remove_extension_callerid2() logs a DEBUG message
instead of an ERROR if the extension you're deleting doesn't exist.
The pjsip_config_wizard calls that function to clean up the config
and has been triggering that annoying error message for years.

Resolves: #582
(cherry picked from commit a433ed0d5a)
2024-03-07 14:17:22 +00:00
Naveen Albert bf6a367955 configure: Rerun bootstrap on modern platform.
The last time configure was run, it was run on a system that
did not enable -std=gnu11 by default, which meant that the
restrict qualifier would not be recognized on certain platforms.
This regenerates the configure files from running bootstrap.sh,
so that these should be recognized on all supported platforms.

Resolves: #586
(cherry picked from commit 54b804fc3b)
2024-03-07 14:17:22 +00:00
Ben Ford ab2163cacb Upgrade bundled pjproject to 2.14.
Fixes: #406

UserNote: Bundled pjproject has been upgraded to 2.14. For more
information on what all is included in this change, check out the
pjproject Github page: https://github.com/pjsip/pjproject/releases

(cherry picked from commit bbfaff33cf)
2024-03-07 14:17:22 +00:00
cmaj db99ad892b app_speech_utils.c: Allow partial speech results.
Adds 'p' option to SpeechBackground() application.
With this option, when the app timeout is reached,
whatever the backend speech engine collected will
be returned as if it were the final, full result.
(This works for engines that make partial results.)

Resolves: #572

UserNote: The SpeechBackground dialplan application now supports a 'p'
option that will return partial results from speech engines that
provide them when a timeout occurs.

(cherry picked from commit 63b5a03538)
2024-03-07 14:17:22 +00:00
Joshua C. Colp ba68bb9221 utils: Make behavior of ast_strsep* match strsep.
Given the scenario of passing an empty string to the
ast_strsep functions the functions would return NULL
instead of an empty string. This is counter to how
strsep itself works.

This change alters the behavior of the functions to
match that of strsep.

Fixes: #565
(cherry picked from commit 167d393c0f)
2024-03-07 14:17:22 +00:00
Mike Bradeen ec69680801 app_chanspy: Add 'D' option for dual-channel audio
Adds the 'D' option to app chanspy that causes the input and output
frames of the spied channel to be interleaved in the spy output frame.
This allows the input and output of the spied channel to be decoded
separately by the receiver.

If the 'o' option is also set, the 'D' option is ignored as the
audio being spied is inherently one direction.

Fixes: #569

UserNote: The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.

(cherry picked from commit 0d17ee6930)
2024-03-07 14:17:22 +00:00
George Joseph 36fd2ac5b6 .github: Update github-script to v7 and fix a rest bug
Need to update the github-script to v7 to squash deprecation
warnings.

Also fixed the API name for github.rest.pulls.requestReviewers.

(cherry picked from commit 0f3f0f3529)
2024-03-07 14:17:22 +00:00
Naveen Albert e2cb63120f app_if: Fix next priority calculation.
Commit fa3922a4d2 fixed
a branching issue but "overshoots" when calculating
the next priority. This fixes that; accompanying
test suite tests have also been extended.

Resolves: #560
(cherry picked from commit 64f9df5e59)
2024-03-07 14:17:22 +00:00
Sean Bright f74eafcf65 res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
The existing code prevented IPv6 addresses from being properly parsed.

Fixes #558

(cherry picked from commit 103a6dc987)
2024-03-07 14:17:22 +00:00
Brad Smith 17757f76fe BuildSystem: Bump autotools versions on OpenBSD.
Bump up to the more commonly used and modern versions of
autoconf and automake.

(cherry picked from commit 921b1d9cbc)
2024-03-07 14:17:22 +00:00
Brad Smith afe0fb310d main/utils: Simplify the FreeBSD ast_get_tid() handling
FreeBSD has had kernel threads for 20+ years.

(cherry picked from commit 57ea2912e2)
2024-03-07 14:17:22 +00:00
Sean Bright 4d09fb191c res_pjsip_session.c: Correctly format SDP connection addresses.
Resolves a regression identified by @justinludwig involving the
rendering of IPv6 addresses in outgoing SDP.

Also updates `media_address` on PJSIP endpoints so that if we are able
to parse the configured value as an IP we store it in a format that we
can directly use later. Based on my reading of the code it appeared
that one could configure `media_address` as:

```
[foo]
type = endpoint
...
media_address = [2001:db8::]
```

And that value would be blindly copied into the outgoing SDP without
regard to its format.

Fixes #541

(cherry picked from commit 0fdaf7fc80)
2024-03-07 14:17:22 +00:00
Sean Bright 347494a130 rtp_engine.c: Correct sample rate typo for L16/44100.
Fixes #555

(cherry picked from commit 2aaf28c95f)
2024-03-07 14:17:22 +00:00
Naveen Albert faf4c3b5d3 manager.c: Fix erroneous reloads in UpdateConfig.
Currently, a reload will always occur if the
Reload header is provided for the UpdateConfig
action. However, we should not be doing a reload
if the header value has a falsy value, per the
documentation, so this makes the reload behavior
consistent with the existing documentation.

Resolves: #551
(cherry picked from commit d50d981543)
2024-03-07 14:17:22 +00:00
Naveen Albert 309ec5e762 res_calendar_icalendar: Print iCalendar error on parsing failure.
If libical fails to parse a calendar, print the error message it provdes.

Resolves: #492
(cherry picked from commit 8eb6a329d6)
2024-03-07 14:17:22 +00:00
Sean Bright 61c78fc48b app_confbridge: Don't emit warnings on valid configurations.
The numeric bridge profile options `internal_sample_rate` and
`maximum_sample_rate` are documented to accept the special values
`auto` and `none`, respectively. While these values currently work,
they also emit warnings when used which could be confusing for users.

In passing, also ensure that we only accept the documented range of
sample rate values between 8000 and 192000.

Fixes #546

(cherry picked from commit f541f8d8c4)
2024-03-07 14:17:22 +00:00
Mike Bradeen 9652a215e8 app_voicemail: add NoOp alembic script to maintain sync
Adding a NoOp alembic script for the voicemail database to maintain
version sync with other branches.

Fixes: #527
(cherry picked from commit f5d622413a)
2024-03-07 14:17:22 +00:00
Naveen Albert e17c4e172f chan_dahdi: Allow MWI to be manually toggled on channels.
This adds a CLI command to manually toggle the MWI status
of a channel, useful for troubleshooting or resetting
MWI devices, similar to the capabilities offered with
SIP messaging to manually control MWI status.

UserNote: The 'dahdi set mwi' now allows MWI on channels
to be manually toggled if needed for troubleshooting.

Resolves: #440
(cherry picked from commit f17910ded5)
2024-03-07 14:17:22 +00:00
PeterHolik 9943bc5f1f chan_rtp.c: MulticastRTP missing refcount without codec option
Fixes: #529
(cherry picked from commit 416d9dfb85)
2024-03-07 14:17:22 +00:00
PeterHolik 07715c4c27 chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
Fixes: asterisk#536
(cherry picked from commit 83d6f0f48b)
2024-03-07 14:17:22 +00:00
Naveen Albert 410909fbfb func_frame_trace: Add CLI command to dump frame queue.
This adds a simple CLI command that can be used for
analyzing all frames currently queued to a channel.

A couple log messages are also adjusted to be more
useful in tracing bridging problems.

Resolves: #533
(cherry picked from commit d075a08d7e)
2024-03-07 14:17:22 +00:00
Asterisk Development Team b05169bcac Update for 20.6.0 2024-01-25 16:19:33 +00:00
Asterisk Development Team f258d08bac Update for 20.6.0-rc2 2024-01-18 16:45:07 +00:00
Naveen Albert 87fb29020b logger: Fix linking regression.
Commit 008731b0a4
caused a regression by resulting in logger.xml
being compiled and linked into the asterisk
binary in lieu of logger.c on certain platforms
if Asterisk was compiled in dev mode.

To fix this, we ensure the file has a unique
name without the extension. Most existing .xml
files have been named differently from any
.c files in the same directory or did not
pose this issue.

channels/pjsip/dialplan_functions.xml does not
pose this issue but is also being renamed
to adhere to this policy.

Resolves: #539
2024-01-17 14:55:02 -07:00
Asterisk Development Team 3ba0470953 Update for 20.6.0-rc1 2024-01-12 18:29:33 +00:00
George Joseph a42c5438e9 Revert "core & res_pjsip: Improve topology change handling."
This reverts commit 315eb551db.

Over the past year, we've had several reports of "topology storms"
occurring where 2 external facing channels connected by one or more
local channels and bridges will get themselves in a state where
they continually send each other topology change requests.  This
usually manifests itself in no-audio calls and a flood of
"Exceptionally long queue length" messages.  It appears that this
commit is the cause so we're reverting it for now until we can
determine a more appropriate solution.

Resolves: #530
(cherry picked from commit 4715c1b11c)
2024-01-12 18:29:20 +00:00
Naveen Albert fd57ddb3e9 menuselect: Use more specific error message.
Instead of using the same error message for
missing dependencies and conflicts, be specific
about what actually went wrong.

Resolves: #520
(cherry picked from commit f22f49e77a)
2024-01-12 18:29:20 +00:00
Maximilian Fridrich b2eb25a88c res_pjsip_nat: Fix potential use of uninitialized transport details
The ast_sip_request_transport_details must be zero initialized,
otherwise this could lead to a SEGV.

Resolves: #509
(cherry picked from commit 3e069f3274)
2024-01-12 18:29:20 +00:00
Naveen Albert d7948f5425 app_if: Fix faulty EndIf branching.
This fixes faulty branching logic for the
EndIf application. Instead of computing
the next priority, which should be done
for false conditionals or ExitIf, we should
simply advance to the next priority.

Resolves: #341
(cherry picked from commit 83a0cb51e5)
2024-01-12 18:29:20 +00:00
Naveen Albert c6b82b19a4 manager.c: Fix regression due to using wrong free function.
Commit 424be34563 introduced
a regression by calling ast_free on memory allocated by
realpath. This causes Asterisk to abort when executing this
function. Since the memory is allocated by glibc, it should
be freed using ast_std_free.

Resolves: #513
(cherry picked from commit b9ed57092f)
2024-01-12 18:29:20 +00:00
Naveen Albert 8a73bac226 config_options.c: Fix truncation of option descriptions.
This increases the format width of option descriptions
to avoid needless truncation for longer descriptions.

Resolves: #428
(cherry picked from commit fcf36a8766)
2024-01-12 18:29:20 +00:00
Naveen Albert 776c2ca6d7 manager.c: Improve clarity of "manager show connected".
Improve the "manager show connected" CLI command
to clarify that the last two columns are permissions
related, not counts, and use sufficient widths
to consistently display these values.

ASTERISK-30143 #close
Resolves: #482

(cherry picked from commit bc53a2a087)
2024-01-12 18:29:20 +00:00
Sean Bright cfe826791e make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
Although `make_xml_documentation`'s `print_dependencies` command was
corrected by the previous fix (#461) for #142, the `create_xml` was
not properly handling `LOCAL_MOD_SUBDIRS` XML documentation.

(cherry picked from commit 91c733bc69)
2024-01-12 18:29:19 +00:00
Naveen Albert 91127a618f general: Fix broken links.
This fixes a number of broken links throughout the
tree, mostly caused by wiki.asterisk.org being replaced
with docs.asterisk.org, which should eliminate the
need for sporadic fixes as in f28047db36.

Resolves: #430
(cherry picked from commit 8f5581b0d0)
2024-01-12 18:29:19 +00:00
George Joseph 98a44b95df MergeApproved.yml: Remove unneeded concurrency
The concurrency parameter on the MergeAndCherryPick job has
been rmeoved.  It was a hold-over from earlier days.

(cherry picked from commit 0005aa2995)
2024-01-12 18:29:19 +00:00
Maximilian Fridrich 8f9200d106 app_dial: Add option "j" to preserve initial stream topology of caller
Resolves: #462

UserNote: The option "j" is now available for the Dial application which
uses the initial stream topology of the caller to create the outgoing
channels.

(cherry picked from commit dcf58ee88f)
2024-01-12 18:29:19 +00:00
George Joseph 9ae72b0a23 ast_coredumper: Increase reliability
Instead of searching for the asterisk binary and the modules in the
filesystem, we now get their locations, along with libdir, from
the coredump itself...

For the binary, we can use `gdb -c <coredump> ... "info proc exe"`.
gdb can print this even without having the executable and symbols.

Once we have the binary, we can get the location of the modules with
`gdb ... "print ast_config_AST_MODULE_DIR`

If there was no result then either it's not an asterisk coredump
or there were no symbols loaded.  Either way, it's not usable.

For libdir, we now run "strings" on the note0 section of the
coredump (which has the shared library -> memory address xref) and
search for "libasteriskssl|libasteriskpj", then take the dirname.

Since we're now getting everything from the coredump, it has to be
correct as long as we're not crossing namespace boundaries like
running asterisk in a docker container but trying to run
ast_coredumper from the host using a shared file system (which you
shouldn't be doing).

There is still a case for using --asterisk-bin and/or --libdir: If
you've updated asterisk since the coredump was taken, the binary,
libraries and modules won't match the coredump which will render it
useless.  If you can restore or rebuild the original files that
match the coredump and place them in a temporary directory, you can
use --asterisk-bin, --libdir, and a new --moddir option to point to
them and they'll be correctly captured in a tarball created
with --tarball-coredumps.  If you also use --tarball-config, you can
use a new --etcdir option to point to what normally would be the
/etc/asterisk directory.

Also addressed many "shellcheck" findings.

Resolves: #445
(cherry picked from commit 44f1522907)
2024-01-12 18:29:19 +00:00
Sean Bright 2c6385a1b3 logger.c: Move LOG_GROUP documentation to dedicated XML file.
The `get_documentation` awk script will only extract the first
DOCUMENTATION block that it finds in a given file. This is by design
(9bc2127) to prevent AMI event documentation from being pulled in to
the core.xml documentation file.

Because of this, the `LOG_GROUP` documentation added in 89709e2 was
not being properly extracted and was missing fom the resulting XML
documentation file. This commit moves the `LOG_GROUP` documentation to
a separate `logger.xml` file.

(cherry picked from commit 0b6e3bc59b)
2024-01-12 18:29:19 +00:00
Matthew Fredrickson 37a89d3cee res_odbc.c: Allow concurrent access to request odbc connections
There are valid scenarios where res_odbc's connection pool might have some dead
or stuck connections while others are healthy (imagine network
elements/firewalls/routers silently timing out connections to a single DB and a
single IP address, or a heterogeneous connection pool connected to potentially
multiple IPs/instances of a replicated DB using a DNS front end for load
balancing and one replica fails).

In order to time out those unhealthy connections without blocking access to
other parts of Asterisk that may attempt access to the connection pool, it would
be beneficial to not lock/block access around the entire pool in
_ast_odbc_request_obj2 while doing potentially blocking operations on connection
pool objects such as the connection_dead() test, odbc_obj_connect(), or by
dereferencing a struct odbc_obj for the last time and triggering a
odbc_obj_disconnect().

This would facilitate much quicker and concurrent timeout of dead connections
via the connection_dead() test, which could block potentially for a long period
of time depending on odbc.ini or other odbc connector specific timeout settings.

This also would make rapid failover (in the clustered DB scenario) much quicker.

This patch changes the locking in _ast_odbc_request_obj2() to not lock around
odbc_obj_connect(), _disconnect(), and connection_dead(), while continuing to
lock around truly shared, non-immutable state like the connection_cnt member and
the connections list on struct odbc_class.

Fixes: #465
(cherry picked from commit bfac3945f6)
2024-01-12 18:29:19 +00:00
Sean Bright 5988da4ec5 res_pjsip_header_funcs.c: Check URI parameter length before copying.
Fixes #477

(cherry picked from commit a2f0d99d9d)
2024-01-12 18:29:19 +00:00
Sean Bright 2f7416711e config.c: Log #exec include failures.
If the script referenced by `#exec` does not exist, writes anything to
stderr, or exits abnormally or with a non-zero exit status, we log
that to Asterisk's error logging channel.

Additionally, write out a warning if the script produces no output.

Fixes #259

(cherry picked from commit 4327ec2907)
2024-01-12 18:29:19 +00:00
Sean Bright 3ed329edc9 make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
If LOCAL_MOD_SUBDIRS contains absolute paths, do not prefix them with
the path to Asterisk's source tree.

Fixes #142

(cherry picked from commit 2293edffd0)
2024-01-12 18:29:19 +00:00
Sean Bright 06d86c41af app_voicemail.c: Completely resequence mailbox folders.
Resequencing is a process that occurs when we open a voicemail folder
and discover that there are gaps between messages (e.g. `msg0000.txt`
is missing but `msg0001.txt` exists). Resequencing involves shifting
the existing messages down so we end up with a sequential list of
messages.

Currently, this process stops after reaching a threshold based on the
message limit (`maxmsg`) configured on the current folder. However, if
`maxmsg` is lowered when a voicemail folder contains more than
`maxmsg + 10` messages, resequencing will not run completely leaving
the mailbox in an inconsistent state.

We now resequence up to the maximum number of messages permitted by
`app_voicemail` (currently hard-coded at 9999 messages).

Fixes #86

(cherry picked from commit 6556a92393)
2024-01-12 18:29:19 +00:00
Naveen Albert 7279d7547f sig_analog: Fix channel leak when mwimonitor is enabled.
When mwimonitor=yes is enabled for an FXO port,
the do_monitor thread will launch mwi_thread if it thinks
there could be MWI on an FXO channel, due to the noise
threshold being satisfied. This, in turns, calls
analog_ss_thread_start in sig_analog. However, unlike
all other instances where __analog_ss_thread is called
in sig_analog, this call path does not properly set
pvt->ss_astchan to the Asterisk channel, which means
that the Asterisk channel is NULL when __analog_ss_thread
starts executing. As a result, the thread exits and the
channel is never properly cleaned up by calling ast_hangup.

This caused issues with do_monitor on incoming calls,
as it would think the channel was still owned even while
receiving events, leading to an infinite barrage of
warning messages; additionally, the channel would persist
improperly.

To fix this, the assignment is added to the call path
where it is missing (which is only used for mwi_thread).
A warning message is also added since previously there
was no indication that __analog_ss_thread was exiting
abnormally. This resolves both the channel leak and the
condition that led to the warning messages.

Resolves: #458
(cherry picked from commit 22e34193ee)
2024-01-12 18:29:19 +00:00
Sean Bright d52c427533 res_rtp_asterisk.c: Update for OpenSSL 3+.
In 5ac5c2b0 we defined `OPENSSL_SUPPRESS_DEPRECATED` to silence
deprecation warnings. This commit switches over to using
non-deprecated API.

(cherry picked from commit 3859b630a7)
2024-01-12 18:29:19 +00:00
Sean Bright b7e66d49b2 alembic: Update list of TLS methods available on ps_transports.
Related to #221 and #222.

Also adds `*.ini` to the `.gitignore` file in ast-db-manage for
convenience.

(cherry picked from commit 0dcf03e844)
2024-01-12 18:29:19 +00:00
Naveen Albert a6d856aba2 func_channel: Expose previously unsettable options.
Certain channel options are not set anywhere or
exposed in any way to users, making them unusable.
This exposes some of these options which make sense
for users to manipulate at runtime.

Resolves: #442
(cherry picked from commit c222343ec6)
2024-01-12 18:29:19 +00:00
Sean Bright f19b74ad31 app.c: Allow ampersands in playback lists to be escaped.
Any function or application that accepts a `&`-separated list of
filenames can now include a literal `&` in a filename by wrapping the
entire filename in single quotes, e.g.:

```
exten = _X.,n,Playback('https://example.com/sound.cgi?a=b&c=d'&hello-world)
```

Fixes #172

UpgradeNote: Ampersands in URLs passed to the `Playback()`,
`Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or
`Queue()` applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the `CONFBRIDGE` dialplan function, or configuring various
features in `confbridge.conf` and `queues.conf`.

(cherry picked from commit f8212d4594)
2024-01-12 18:29:19 +00:00
Sean Bright 989e61890a uri.c: Simplify ast_uri_make_host_with_port()
(cherry picked from commit ff012323e8)
2024-01-12 18:29:19 +00:00
Sean Bright 69a19aabd8 func_curl.c: Remove CURLOPT() plaintext documentation.
I assume this was missed when initially converting to XML
documentation and we've been kicking the can down the road since.

(cherry picked from commit 9e1a60727e)
2024-01-12 18:29:19 +00:00
Sean Bright fe92d09361 res_http_websocket.c: Set hostname on client for certificate validation.
Additionally add a `assert()` to in the TLS client setup code to
ensure that hostname is set when it is supposed to be.

Fixes #433

(cherry picked from commit f2961f048d)
2024-01-12 18:29:19 +00:00
Sean Bright eb48273bdf live_ast: Add astcachedir to generated asterisk.conf.
`astcachedir` (added in b0842713) was not added to `live_ast` so
continued to point to the system `/var/cache` directory instead of the
one in the live environment.

(cherry picked from commit 978d09fc35)
2024-01-12 18:29:19 +00:00
George Joseph 3127baec78 SECURITY.md: Update with correct documentation URL
(cherry picked from commit d10d4d9ddd)
2024-01-12 18:29:19 +00:00
Naveen Albert 4d928ee975 func_lock: Add missing see-also refs to documentation.
Resolves: #423
(cherry picked from commit 12b353eae0)
2024-01-12 18:29:19 +00:00
Matthew Fredrickson eac9ad69a8 app_followme.c: Grab reference on nativeformats before using it
Fixes a crash due to a lack of proper reference on the nativeformats
object before passing it into ast_request().  Also found potentially
similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c

Fixes: #388
(cherry picked from commit 275f7911b5)
2024-01-12 18:29:19 +00:00
Naveen Albert 52388f11f8 configs: Improve documentation for bandwidth in iax.conf.
This improves the documentation for the bandwidth setting
in iax.conf by making it clearer what the ramifications
of this setting are. It also changes the sample default
from low to high, since only high is compatible with good
codecs that people will want to use in the vast majority
of cases, and this is a common gotcha that trips up new users.

Resolves: #425
(cherry picked from commit 1f19227eab)
2024-01-12 18:29:19 +00:00
Naveen Albert 0007625ad0 logger: Add channel-based filtering.
This adds the ability to filter console
logging by channel or groups of channels.
This can be useful on busy systems where
an administrator would like to analyze certain
calls in detail. A dialplan function is also
included for the purpose of assigning a channel
to a group (e.g. by tenant, or some other metric).

ASTERISK-30483 #close

Resolves: #242

UserNote: The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.

(cherry picked from commit a0fc8d1b5f)
2024-01-12 18:29:19 +00:00
Sean Bright c2680f63c5 chan_iax2.c: Don't send unsanitized data to the logger.
This resolves an issue where non-printable characters could be sent to
the console/log files.

(cherry picked from commit d2afb10eed)
2024-01-12 18:29:19 +00:00
George Joseph 1d87c27cab codec_ilbc: Disable system ilbc if version >= 3.0.0
Fedora 37 started shipping ilbc 3.0.4 which we don't yet support.
configure.ac now checks the system for "libilbc < 3" instead of
just "libilbc".  If true, the system version of ilbc will be used.
If not, the version included at codecs/ilbc will be used.

Resolves: #84
(cherry picked from commit d819a6bccb)
2024-01-12 18:29:19 +00:00
Sean Bright da35b6a244 resource_channels.c: Explicit codec request when creating UnicastRTP.
Fixes #394

(cherry picked from commit a83c761c95)
2024-01-12 18:29:19 +00:00
Sean Bright fb7d39db6d doc: Update IP Quality of Service links.
Fixes #328

(cherry picked from commit 26918d05f4)
2024-01-12 18:29:19 +00:00
George Joseph 94f931a6d7 chan_pjsip: Add PJSIPHangup dialplan app and manager action
See UserNote below.

Exposed the existing Hangup AMI action in manager.c so we can use
all of it's channel search and AMI protocol handling without
duplicating that code in dialplan_functions.c.

Added a lookup function to res_pjsip.c that takes in the
string represenation of the pjsip_status_code enum and returns
the actual status code.  I.E.  ast_sip_str2rc("DECLINE") returns
603.  This allows the caller to specify PJSIPHangup(decline) in
the dialplan, just like Hangup(call_rejected).

Also extracted the XML documentation to its own file since it was
almost as large as the code itself.

UserNote: A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.

(cherry picked from commit cd77953172)
2024-01-12 18:29:19 +00:00
Sean Bright f96d7ef7b5 chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
When IAX2 debugging was enabled (`iax2 set debug on`), if the last IE
in a frame was one that may not have any data - such as the CALLTOKEN
IE in an NEW request - it was not getting displayed.

(cherry picked from commit 0e126b3841)
2024-01-12 18:29:19 +00:00
Naveen Albert e75aebc9bc chan_dahdi: Warn if nonexistent cadence is requested.
If attempting to ring a channel using a nonexistent cadence,
emit a warning, before falling back to the default cadence.

Resolves: #409
(cherry picked from commit 4b9a4483fc)
2024-01-12 18:29:19 +00:00
Holger Hans Peter Freyther 7699af00e1 stasis: Update the snapshot after setting the redirect
The previous commit added the caller_rdnis attribute. Make it
avialble during a possible ChanngelHangupRequest.

(cherry picked from commit 56733c73b4)
2024-01-12 18:29:19 +00:00
Holger Hans Peter Freyther 28f52d35f3 ari: Provide the caller ID RDNIS for the channels
Provide the caller ID RDNIS when available. This will allow an
application to follow the redirect.

(cherry picked from commit 157389bc59)
2024-01-12 18:29:19 +00:00
Brad Smith 089ddaaaed main/utils: Implement ast_get_tid() for OpenBSD
Implement the ast_get_tid() function for OpenBSD. OpenBSD supports
getting the TID via getthrid().

(cherry picked from commit e7943dd4d9)
2024-01-12 18:29:19 +00:00
Brad Smith fb3067dd71 res_rtp_asterisk.c: Fix runtime issue with LibreSSL
The module will fail to load. Use proper function DTLS_method() with LibreSSL.

(cherry picked from commit 65d38c8104)
2024-01-12 18:29:19 +00:00
Naveen Albert 6a4fe8bdab app_directory: Add ADSI support to Directory.
This adds optional ADSI support to the Directory
application, which allows callers with ADSI CPE
to navigate the Directory system significantly
faster than is possible using the audio prompts.
Callers can see the directory name (and optionally
extension) on their screenphone and confirm or
reject a match immediately rather than waiting
for it to be spelled out, enhancing usability.

Resolves: #356
(cherry picked from commit 5046620fa3)
2024-01-12 18:29:19 +00:00
Naveen Albert b47a403b80 core_local: Fix local channel parsing with slashes.
Currently, trying to call a Local channel with a slash
in the extension will fail due to the parsing of characters
after such a slash as being dial modifiers. Additionally,
core_local is inconsistent and incomplete with
its parsing of Local dial strings in that sometimes it
uses the first slash and at other times it uses the last.

For instance, something like DAHDI/5 or PJSIP/device
is a perfectly usable extension in the dialplan, but Local
channels in particular prevent these from being called.

This creates inconsistent behavior for users, since using
a slash in an extension is perfectly acceptable, and using
a Goto to accomplish this works fine, but if specified
through a Local channel, the parsing prevents this.

This fixes this by explicitly parsing options from the
last slash in the extension, rather than the first one,
which doesn't cause an issue for extensions with slashes.

ASTERISK-30013 #close

Resolves: #248
(cherry picked from commit 2191a0d33f)
2024-01-12 18:29:19 +00:00
Mark Murawski 3b70dfcced Remove files that are no longer updated
Fixes: #360
(cherry picked from commit 2ed8daa3cb)
2024-01-12 18:29:19 +00:00
Naveen Albert b5eb5e19c3 app_voicemail: Add AMI event for mailbox PIN changes.
This adds an AMI event that is emitted whenever a
mailbox password is successfully changed, allowing
AMI consumers to process these.

UserNote: The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.

Resolves: #398
(cherry picked from commit cf9d7fa9f6)
2024-01-12 18:29:19 +00:00
Sean Bright 3ded0c95fc app_queue.c: Emit unpause reason with PauseQueueMember event.
Fixes #395

(cherry picked from commit fb937d1d89)
2024-01-12 18:29:19 +00:00
George Joseph e2f246c067 bridge_simple: Suppress unchanged topology change requests
In simple_bridge_join, we were sending topology change requests
even when the new and old topologies were the same.  In some
circumstances, this can cause unnecessary re-invites and even
a re-invite flood.  We now suppress those.

Resolves: #384
(cherry picked from commit f301d4559e)
2024-01-12 18:29:19 +00:00
Naveen Albert 120dbba2b3 res_pjsip: Include cipher limit in config error message.
If too many ciphers are specified in the PJSIP config,
include the maximum number of ciphers that may be
specified in the user-facing error message.

Resolves: #396
(cherry picked from commit 8d9d0d3738)
2024-01-12 18:29:19 +00:00
Mike Bradeen 0b4427d621 res_speech: allow speech to translate input channel
* Allow res_speech to translate the input channel if the
  format is translatable to a format suppored by the
  speech provider.

Resolves: #129

UserNote: res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.

(cherry picked from commit c3e2bff36e)
2024-01-12 18:29:19 +00:00
Sean Bright ea74f942ce res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
Fixes #386

(cherry picked from commit deeb1acffe)
2024-01-12 18:29:19 +00:00
Sean Bright 58f78d2d0e res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
Fixes #376

(cherry picked from commit d4e4942cf5)
2024-01-12 18:29:19 +00:00
George Joseph 027c748e57 api.wiki.mustache: Fix indentation in generated markdown
The '*' list indicator for default values and allowable values for
path, query and POST parameters need to be indented 4 spaces
instead of 2.

Should resolve issue 38 in the documentation repo.

(cherry picked from commit b619b64137)
2024-01-12 18:29:19 +00:00
Sean Bright 93697458a6 pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
Per RFC8827:

    Implementations MUST NOT implement DTLS renegotiation and MUST
    reject it with a "no_renegotiation" alert if offered.

So we disable it when webrtc=yes is set.

Fixes #378

UpgradeNote: The dtls_rekey will be disabled if webrtc support is
requested on an endpoint. A warning will also be emitted.

(cherry picked from commit 9d329da346)
2024-01-12 18:29:19 +00:00
Samuel Olaechea 59181860ec configs: Fix typo in pjsip.conf.sample.
(cherry picked from commit ebc78a83be)
2024-01-12 18:29:19 +00:00
George Joseph e52175341e res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
Commit f66f77f last year prevents the res_pjsip_exten_state and
res_pjsip_mwi modules from unloading due to possible pjproject
asserts if the modules are reloaded. A side effect of the
implementation is that the taskprocessors these modules use aren't
being released. When asterisk is doing a graceful shutdown, it
waits AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT seconds for all
taskprocessors to stop but since those 2 modules don't release
theirs, the shutdown hangs for that amount of time.

This change allows the modules to be unloaded and their resources to
be released when ast_shutdown_final is true.

Resolves: #379
(cherry picked from commit 9efc4bdfbc)
2024-01-12 18:29:19 +00:00
sungtae kim faf046ea86 res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
This commit introduces an extension to the endpoint and relevant
resource sizes for PJSIP, transitioning from its current 40-character
constraint to a more versatile 255-character capacity. This enhancement
significantly overcomes limitations related to domain qualification and
practical usage, ultimately delivering improved functionality. In
addition, it includes adjustments to accommodate the expanded realm size
within the ARI, specifically enhancing the maximum realm length.

Resolves: #345

UserNote: With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.

UpgradeNote: As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.

(cherry picked from commit f89e56c178)
2024-01-12 18:29:19 +00:00
George Joseph 95656c409d .github: PRSubmitActions: Fix adding reviewers to PR
(cherry picked from commit be1e83a6ac)
2024-01-12 18:29:19 +00:00
George Joseph 13f2d126fd .github: New PR Submit workflows
The workflows that get triggered when PRs are submitted or updated
have been replaced with ones that are more secure and have
a higher level of parallelism.

(cherry picked from commit 289aa1840e)
2024-01-12 18:29:19 +00:00
George Joseph 2cb140f801 .github: New PR Submit workflows
The workflows that get triggered when PRs are submitted or updated
have been replaced with ones that are more secure and have
a higher level of parallelism.

(cherry picked from commit 0c1c6e9ada)
2024-01-12 18:29:19 +00:00
Mike Bradeen 0f2eb00a52 res_stasis: signal when new command is queued
res_statsis's app loop sleeps for up to .2s waiting on input
to a channel before re-checking the command queue. This can
cause delays between channel setup and bridge.

This change is to send a SIGURG on the sleeping thread when
a new command is enqueued. This exits the sleeping thread out
of the ast_waitfor() call triggering the new command being
processed on the channel immediately.

Resolves: #362

UserNote: Call setup times should be significantly improved
when using ARI.

(cherry picked from commit 79220e3f0c)
2024-01-12 18:29:19 +00:00
Holger Hans Peter Freyther 1f7e9bfad5 ari/stasis: Indicate progress before playback on a bridge
Make it possible to start a playback and the calling party
to receive audio on a bridge before the call is connected.

Model the implementation after play_on_channel and deliver a
AST_CONTROL_PROGRESS before starting the playback.

For a PJSIP channel this will result in sending a SIP 183
Session Progress.

(cherry picked from commit 1171dcee02)
2024-01-12 18:29:18 +00:00
Sean Bright d7a01afa50 func_curl.c: Ensure channel is locked when manipulating datastores.
(cherry picked from commit acb2348f90)
2024-01-12 18:29:18 +00:00
George Joseph fd487b48a3 .github: Fix job prereqs in PROpenedUpdated
(cherry picked from commit 20398e8e95)
2024-01-12 18:29:18 +00:00
George Joseph e1ce35c3f5 .github: Block PR tests until approved
(cherry picked from commit e9abf11a26)
2024-01-12 18:29:18 +00:00
Joshua C. Colp b7cc7e804b Update config.yml
(cherry picked from commit ef7b0f4c3b)
2024-01-12 18:29:18 +00:00
George Joseph f0edab9f3e logger.h: Add ability to change the prefix on SCOPE_TRACE output
You can now define the _TRACE_PREFIX_ macro to change the
default trace line prefix of "file:line function" to
something else.  Full documentation in logger.h.

(cherry picked from commit b52e07ee1b)
2024-01-12 18:29:18 +00:00
George Joseph 393d8cb947 Add libjwt to third-party
The current STIR/SHAKEN implementation is not currently usable due
to encryption issues. Rather than trying to futz with OpenSSL and
the the current code, we can take advantage of the existing
capabilities of libjwt but we first need to add it to the
third-party infrastructure already in place for jansson and
pjproject.

A few tweaks were also made to the third-party infrastructure as
a whole.  The jansson "dest" install directory was renamed "dist"
to better match convention, and the third-party Makefile was updated
to clean all product directories not just the ones currently in
use.

Resolves: #349
(cherry picked from commit a9d4175e1d)
2024-01-12 18:29:18 +00:00
Mike Bradeen 0df2848147 res_pjsip: update qualify_timeout documentation with DNS note
The documentation on qualify_timeout does not explicitly state that the timeout
includes any time required to perform any needed DNS queries on the endpoint.

If the OPTIONS response is delayed due to the DNS query, it can still render an
endpoint as Unreachable if the net time is enough for qualify_timeout to expire.

Resolves: #352
(cherry picked from commit 933490b758)
2024-01-12 18:29:18 +00:00
Naveen Albert c06fd10f0a chan_dahdi: Clarify scope of callgroup/pickupgroup.
Internally, chan_dahdi only applies callgroup and
pickupgroup to FXO signalled channels, but this is
not documented anywhere. This is now documented in
the sample config, and a warning is emitted if a
user tries configuring these settings for channel
types that do not support these settings, since they
will not have any effect.

Resolves: #294
(cherry picked from commit 945babf25c)
2024-01-12 18:29:18 +00:00
Bastian Triller 7a1208410a func_json: Fix crashes for some types
This commit fixes crashes in JSON_DECODE() for types null, true, false
and real numbers.

In addition it ensures that a path is not deeper than 32 levels.

Also allow root object to be an array.

Add unit tests for above cases.

(cherry picked from commit e6d5b8d8cf)
2024-01-12 18:29:18 +00:00
Mike Bradeen c47a386cf2 res_speech_aeap: add aeap error handling
res_speech_aeap previously did not register an error handler
with aeap, so it was not notified of a disconnect. This resulted
in SpeechBackground never exiting upon a websocket disconnect.

Resolves: #303
(cherry picked from commit 8c934fb7ed)
2024-01-12 18:29:18 +00:00
Naveen Albert 0a834c8d75 app_voicemail: Disable ADSI if unavailable.
If ADSI is available on a channel, app_voicemail will repeatedly
try to use ADSI, even if there is no CPE that supports it. This
leads to many unnecessary delays during the session. If ADSI is
available but ADSI setup fails, we now disable it to prevent
further attempts to use ADSI during the session.

Resolves: #354
(cherry picked from commit c04923fcda)
2024-01-12 18:29:18 +00:00
Eduardo 7a2305a08b codec_builtin: Use multiples of 20 for maximum_ms
Some providers require a multiple of 20 for the maxptime or fail to complete calls,
e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used.

Resolves: #260
(cherry picked from commit 11d87713eb)
2024-01-12 18:29:18 +00:00
George Joseph 9afa54b3b3 lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS.
Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead
to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just
causes the lock calls to loop over trylock in 200us intervals until
the lock is obtained and spits out log messages if it takes more
than 5 seconds.  From a code perspective, the only reason they were
tied together was for logging.  So... The ifdefs in lock.c were
refactored to allow DETECT_DEADLOCKS to be enabled without
also enabling DEBUG_THREADS.

Resolves: #321

UserNote: You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS.  This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.

(cherry picked from commit 7e2243f9e1)
2024-01-12 18:29:18 +00:00
George Joseph 97674e1d20 asterisk.c: Use the euid's home directory to read/write cli history
The CLI .asterisk_history file is read from/written to the directory
specified by the HOME environment variable. If the root user starts
asterisk with the -U/-G options, or with runuser/rungroup set in
asterisk.conf, the asterisk process is started as root but then it
calls setuid/setgid to set the new user/group. This does NOT reset
the HOME environment variable to the new user's home directory
though so it's still left as "/root". In this case, the new user
will almost certainly NOT have access to read from or write to the
history file.

* Added function process_histfile() which calls
  getpwuid(geteuid()) and uses pw->dir as the home directory
  instead of the HOME environment variable.
* ast_el_read_default_histfile() and ast_el_write_default_histfile()
  have been modified to use the new process_histfile()
  function.

Resolves: #337
(cherry picked from commit 07cf37531a)
2024-01-12 18:29:18 +00:00
Tinet-mucw a86376fe68 res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
From the gdb information, ast_websocket_read reads a message successfully,
then transport_read is called in the serializer. During execution of pjsip_transport_down,
ws_session->stream->fd is closed; ast_websocket_read encounters an error and exits the while loop.
After executing transport_shutdown, the transport's reference count becomes 0, causing a crash when sending SIP messages.
This was due to pjsip_transport_dec_ref executing earlier than pjsip_rx_data_clone, leading to this issue.
In websocket_cb executeing pjsip_transport_add_ref, this we now ensure the transport is not destroyed while in the loop.

Resolves: asterisk#299
(cherry picked from commit edc674a6ca)
2024-01-12 18:29:18 +00:00
Mike Bradeen 3b61583c20 cel: add publish user event helper
Add a wrapper function around ast_cel_publish_event that
packs event and extras into a blob before publishing

Resolves:#330
(cherry picked from commit 248b92563c)
2024-01-12 18:29:18 +00:00
Naveen Albert 3fef46e776 chan_console: Fix deadlock caused by unclean thread exit.
To terminate a console channel, stop_stream causes pthread_cancel
to make stream_monitor exit. However, commit 5b8fea93d1
added locking to this function which results in deadlock due to
the stream_monitor thread being killed while it's holding the pvt lock.

To resolve this, a flag is now set and read to indicate abort, so
the use of pthread_cancel and pthread_kill can be avoided altogether.

Resolves: #308
(cherry picked from commit 71215561d8)
2024-01-12 18:29:18 +00:00
George Joseph d5a285c8f2 file.c: Add ability to search custom dir for sounds
To better co-exist with sounds files that may be managed by
packages, custom sound files may now be placed in
AST_DATA_DIR/sounds/custom instead of the standard
AST_DATA_DIR/sounds/<lang> directory.  If the new
"sounds_search_custom_dir" option in asterisk.conf is set
to "true", asterisk will search the custom directory for sounds
files before searching the standard directory.  For performance
reasons, the "sounds_search_custom_dir" defaults to "false".

Resolves: #315

UserNote: A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/<lang> directory.

(cherry picked from commit 4493d2b2fc)
2024-01-12 18:29:18 +00:00
Naveen Albert accedd4c3e chan_iax2: Improve authentication debugging.
Improves and adds some logging to make it easier
for users to debug authentication issues.

Resolves: #286
(cherry picked from commit 833ee80789)
2024-01-12 18:29:18 +00:00
Vitezslav Novy c275ee9e54 res_rtp_asterisk: fix wrong counter management in ioqueue objects
In function  rtp_ioqueue_thread_remove counter in ioqueue object is not decreased
which prevents unused ICE TURN threads from being removed.

Resolves: #301
(cherry picked from commit 5179f1af24)
2024-01-12 18:29:18 +00:00
George Joseph 172602d30e make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
The previous behavior of make_buildopts_h was to not add the
non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS,
REF_DEBUG, etc. to the buildopts.h file because "it caused
ccache to invalidate files and extended compile times". They're
only defined by passing them on the gcc command line with '-D'
options.   In practice, including them in the include file rarely
causes any impact because the only time ccache cares is if you
actually change an option so the hit occurrs only once after
you change it.

OK so why would we want to include them?  Many IDEs follow the
include files to resolve defines and if the options aren't in an
include file, it can cause the IDE to mark blocks of "ifdeffed"
code as unused when they're really not.

So...

* Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H
  which tells make_buildopts_h to include the non-ABI-breaking
  flags in buildopts.h as well as the ABI-breaking ones. The default
  is disabled to preserve current behavior.  As before though,
  only the ABI-breaking flags appear in AST_BUILDOPTS and only
  those are used to calculate AST_BUILDOPT_SUM.
  A new AST_BUILDOPT_ALL define was created to capture all of the
  flags.

* make_version_c was streamlined to use buildopts.h and also to
  create asterisk_build_opts_all[] and ast_get_build_opts_all(void)

* "core show settings" now shows both AST_BUILDOPTS and
  AST_BUILDOPTS_ALL.

UserNote: The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.

(cherry picked from commit 06da7b342e)
2024-01-12 18:29:18 +00:00
Mike Bradeen 58a37523dd func_periodic_hook: Add hangup step to avoid timeout
func_periodic_hook does not hangup after playback, relying on hangup
which keeps the channel alive longer than necessary.

Resolves: #325
(cherry picked from commit dd817c2708)
2024-01-12 18:29:18 +00:00
Sean Bright d9dc7b66e4 res_stasis_recording.c: Save recording state when unmuted.
Fixes #322

(cherry picked from commit 3fafd7c0b7)
2024-01-12 18:29:18 +00:00
Mike Bradeen b68370ba9c res_speech_aeap: check for null format on response
* Fixed issue in res_speech_aeap when unable to provide an
  input format to check against.

(cherry picked from commit b53e946b59)
2024-01-12 18:29:18 +00:00
George Joseph 9ee35c259a func_periodic_hook: Don't truncate channel name
func_periodic_hook was truncating long channel names which
causes issues when you need to run other dialplan functions/apps
on the channel.

Resolves: #319
(cherry picked from commit 9e2433f73f)
2024-01-12 18:29:18 +00:00
George Joseph 52929efb83 safe_asterisk: Change directory permissions to 755
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
(cherry picked from commit c929146c61)
2024-01-12 18:29:18 +00:00
Maximilian Fridrich 38598701da chan_rtp: Implement RTP glue for UnicastRTP channels
Resolves: #298

UserNote: The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.

(cherry picked from commit 98ffcfebda)
2024-01-12 18:29:18 +00:00
Jaco Kroon bc72c76891 app_queue: periodic announcement configurable start time.
This newly introduced periodic-announce-startdelay makes it possible to
configure the initial start delay of the first periodic announcement
after which periodic-announce-frequency takes over.

UserNote: Introduce a new queue configuration option called
'periodic-announce-startdelay' which will vary the normal (historic)
behavior of starting the periodic announcement cycle at
periodic-announce-frequency seconds after entering the queue to start
the periodic announcement cycle at period-announce-startdelay seconds
after joining the queue.  The default behavior if this config option is
not set remains unchanged.

Signed-off-by: Jaco Kroon <jaco@uls.co.za>
(cherry picked from commit a4cb63e231)
2024-01-12 18:29:18 +00:00
Joshua C. Colp 6521066fbe variables: Add additional variable dialplan functions.
Using the Set dialplan application does not actually
delete channel or global variables. Instead the
variables are set to an empty value.

This change adds two dialplan functions,
GLOBAL_DELETE and DELETE which can be used to
delete global and channel variables instead
of just setting them to empty.

There is also no ability within the dialplan to
determine if a global or channel variable has
actually been set or not.

This change also adds two dialplan functions,
GLOBAL_EXISTS and VARIABLE_EXISTS which can be
used to determine if a global or channel variable
has been set or not.

Resolves: #289

UserNote: Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.

(cherry picked from commit abe4f62554)
2024-01-12 18:29:18 +00:00
George Joseph c2d3ad489d Restore CHANGES and UPGRADE.txt to allow cherry-picks to work 2024-01-12 11:04:57 -07:00
Asterisk Development Team 4988f9131f Update for 20.5.2 2023-12-20 16:23:39 +00:00
George Joseph 32b57067a6 res_rtp_asterisk: Fix regression issues with DTLS client check
* Since ICE candidates are used for the check and pjproject is
  required to use ICE, res_rtp_asterisk was failing to compile
  when pjproject wasn't available.  The check is now wrapped
  with an #ifdef HAVE_PJPROJECT.

* The rtp->ice_active_remote_candidates container was being
  used to check the address on incoming packets but that
  container doesn't contain peer reflexive candidates discovered
  during negotiation. This was causing the check to fail
  where it shouldn't.  We now check against pjproject's
  real_ice->rcand array which will contain those candidates.

* Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where
  we weren't zeroing out sin->sin_zero before returning.  This
  was causing ast_sockaddr_cmp() to always return false when
  one of the inputs was converted from a pj_sockaddr, even
  if both inputs had the same address and port.

Resolves: #500
Resolves: #503
Resolves: #505
2023-12-20 08:47:21 -07:00
Asterisk Development Team f7a8ac086d Update for 20.5.1 2023-12-14 20:02:23 +00:00
Gitea b6122db51d res_pjsip_header_funcs: Duplicate new header value, don't copy.
When updating an existing header the 'update' code incorrectly
just copied the new value into the existing buffer. If the
new value exceeded the available buffer size memory outside
of the buffer would be written into, potentially causing
a crash.

This change makes it so that the 'update' now duplicates
the new header value instead of copying it into the existing
buffer.
2023-12-14 12:00:51 -07:00
Mike Bradeen 48e435f84f res_pjsip: disable raw bad packet logging
Add patch to split the log level for invalid packets received on the
signaling port.  The warning regarding the packet will move to level 2
so that it can still be displayed, while the raw packet will be at level
4.
2023-12-14 12:00:38 -07:00
George Joseph e46cc258ab res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
When ICE is in use, we can prevent a possible DOS attack by allowing
DTLS protocol messages (client hello, etc) only from sources that
are in the active remote candidates list.

Resolves: GHSA-hxj9-xwr8-w8pq
2023-12-14 12:00:29 -07:00
Ben Ford d44f07244a manager.c: Prevent path traversal with GetConfig.
When using AMI GetConfig, it was possible to access files outside of the
Asterisk configuration directory by using filenames with ".." and "./"
even while live_dangerously was not enabled. This change resolves the
full path and ensures we are still in the configuration directory before
attempting to access the file.
2023-12-14 12:00:19 -07:00
Asterisk Development Team c94e2b2000 Update for 20.5.0 2023-10-18 17:10:25 +00:00
Asterisk Development Team a2fb6d4dd4 Update for 20.5.0-rc1 2023-09-06 16:56:09 +00:00
George Joseph f7fd2c8e20 ari-stubs: Fix more local anchor references
Also allow CreateDocs job to be run manually with default branches.

(cherry picked from commit a64718c32c)
2023-09-06 16:56:00 +00:00
George Joseph 1d9ea90058 ari-stubs: Fix more local anchor references
Also allow CreateDocs job to be run manually with default branches.

(cherry picked from commit b7dae87d1d)
2023-09-06 16:56:00 +00:00
George Joseph e5a653622a ari-stubs: Fix broken documentation anchors
All of the links that reference page anchors with capital letters in
the ids (#Something) have been changed to lower case to match the
anchors that are generated by mkdocs.

(cherry picked from commit c06f938851)
2023-09-06 16:56:00 +00:00
Bastian Triller 9d41064bc0 res_pjsip_session: Send Session Interval too small response
Handle session interval lower than endpoint's configured minimum timer
when sending first answer. Timer setting is checked during this step and
needs to handled appropriately.
Before this change, no response was sent at all. After this change a
response with 422 Session Interval too small is sent to UAC.

(cherry picked from commit 0c0b99c5a1)
2023-09-06 16:56:00 +00:00
George Joseph abc84c6618 .github: Update workflow-application-token-action to v2
(cherry picked from commit ddbc56505e)
2023-09-06 16:56:00 +00:00
Naveen Albert 7959c64792 app_dial: Fix infinite loop when sending digits.
If the called party hangs up while digits are being
sent, -1 is returned to indicate so, but app_dial
was not checking the return value, resulting in
the hangup being lost and looping forever until
the caller manually hangs up the channel. We now
abort if digit sending fails.

ASTERISK-29428 #close

Resolves: #281
(cherry picked from commit 4542ffe5d5)
2023-09-06 16:56:00 +00:00
Mike Bradeen 7353462be3 app_voicemail: Fix for loop declarations
Resolve for loop initial declarations added in cli changes.

Resolves: #275
(cherry picked from commit 36b749ddf8)
2023-09-06 16:56:00 +00:00
George Joseph 4dcf73b6df alembic: Fix quoting of the 100rel column
Add quoting around the ps_endpoints 100rel column in the ALTER
statements.  Although alembic doesn't complain when generating
sql statements, postgresql does (rightly so).

Resolves: #274
(cherry picked from commit cb9223cdb9)
2023-09-06 16:56:00 +00:00
Naveen Albert 2d1e4aed4f pbx.c: Fix gcc 12 compiler warning.
Resolves: #277
(cherry picked from commit 5077301de6)
2023-09-06 16:56:00 +00:00
zhengsh 6ef2389e9b app_audiosocket: Fixed timeout with -1 to avoid busy loop.
Resolves: asterisk#234
(cherry picked from commit afe461419e)
2023-09-06 16:56:00 +00:00
George Joseph c3f4b56886 download_externals: Fix a few version related issues
* Fixed issue with the script not parsing the new tag format for
  certified releases.  The format changed from certified/18.9-cert5
  to certified-18.9-cert5.

* Fixed issue where the asterisk version wasn't being considered
  when looking for cached versions.

Resolves: #263
(cherry picked from commit 715594767d)
2023-09-06 16:56:00 +00:00
Maximilian Fridrich b2cd0583c9 main/refer.c: Fix double free in refer_data_destructor + potential leak
Resolves: #267
(cherry picked from commit fcdffe0074)
2023-09-06 16:56:00 +00:00
Naveen Albert 534269e3e6 sig_analog: Add Called Subscriber Held capability.
This adds support for Called Subscriber Held for FXS
lines, which allows users to go on hook when receiving
a call and resume the call later from another phone on
the same line, without disconnecting the call. This is
a convenience mechanism that most real PSTN telephone
switches support.

ASTERISK-30372 #close

Resolves: #240

UserNote: Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station  user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.

(cherry picked from commit 52283301e8)
2023-09-06 16:56:00 +00:00
Matthew Fredrickson 8bf9b1d966 app_macro: Fix locking around datastore access
app_macro sometimes would crash due to datastore list corruption on the
channel because of lack of locking around find and create process for
the macro datastore. This patch locks the channel lock prior to protect
against this problem.

Resolves: #265
(cherry picked from commit 9a28531362)
2023-09-06 16:56:00 +00:00
Matthew Fredrickson 65082ae112 Revert "app_stack: Print proper exit location for PBXless channels."
This reverts commit 617dad4cba.

apps/app_stack.c: Revert buggy gosub patch

This seems to break the case when a predial macro calls a gosub.
When the gosub calls return, the Return function outputs:

app_stack.c:423 return_exec: Return without Gosub: stack is empty

This returns -1 to the calling macro, which returns to app_dial
and causes the call to hangup instead of proceeding with the macro
that invoked the gosub.

Resolves: #253
(cherry picked from commit c8f2698ec6)
2023-09-06 16:56:00 +00:00
George Joseph b145430957 .github: Use generic releaser
(cherry picked from commit d9494ca392)
2023-09-06 16:56:00 +00:00
Jason D. McCormick f86f5fd809 install_prereq: Fix dependency install on aarch64.
Fixes dependency solutions in install_prereq for Debian aarch64
platforms. install_prereq was attempting to forcibly install 32-bit
armhf packages due to the aptitude search for dependencies.

Resolves: #37
(cherry picked from commit fc57dca50d)
2023-09-06 16:56:00 +00:00
MikeNaso 714b9e8b21 res_pjsip.c: Set contact_user on incoming call local Contact header
If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.

Resolves: #226
(cherry picked from commit b286d1cca2)
2023-09-06 16:56:00 +00:00
Sean Bright cce02a9d60 extconfig: Allow explicit DB result set ordering to be disabled.
Added a new boolean configuration flag -
`order_multi_row_results_by_initial_column` - to both res_pgsql.conf
and res_config_odbc.conf that allows the administrator to disable the
explicit `ORDER BY` that was previously being added to all generated
SQL statements that returned multiple rows.

Fixes: #179
(cherry picked from commit 29eb4fe620)
2023-09-06 16:55:59 +00:00
George Joseph 44d7349958 rest-api: Run make ari-stubs
An earlier cherry-pick that involved rest-api somehow didn't include
a comment change in res/ari/resource_endpoints.h.  This commit
corrects that.  No changes other than the comment.

(cherry picked from commit 64b29be702)
2023-09-06 16:55:59 +00:00
Naveen Albert 9367926380 res_pjsip_header_funcs: Make prefix argument optional.
The documentation for PJSIP_HEADERS claims that
prefix is optional, but in the code it is actually not.
However, there is no inherent reason for this, as users
may want to retrieve all header names, not just those
beginning with a certain prefix.

This makes the prefix optional for this function,
simply fetching all header names if not specified.
As a result, the documentation is now correct.

Resolves: #230

UserNote: The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.

(cherry picked from commit 8be503b663)
2023-09-06 16:55:59 +00:00
George Joseph 5e3652e738 pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
The default is 32 with 8 being used by pjproject itself.  Recent
commits have put us over the limit resulting in assertions in
pjproject.  Since this value is used in invites, dialogs,
transports and subscriptions as well as the global pjproject
endpoint, we don't want to increase it too much.

Resolves: #255
(cherry picked from commit caf51592e7)
2023-09-06 16:55:59 +00:00
Joshua C. Colp 592c85b8de manager: Tolerate stasis messages with no channel snapshot.
In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.

(cherry picked from commit 07d798875d)
2023-09-06 16:55:59 +00:00
Maximilian Fridrich 68fc422247 core/ari/pjsip: Add refer mechanism
This change adds support for refers that are not session based. It
includes a refer implementation for the PJSIP technology which results
in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
triggered using the new ARI endpoint `/endpoints/refer`.

Resolves: #71

UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.

(cherry picked from commit 8824b845c4)
2023-09-06 16:55:59 +00:00
Naveen Albert 65a2bca074 chan_dahdi: Allow autoreoriginating after hangup.
Currently, if an FXS channel is still off hook when
all calls on the line have hung up, the user is provided
reorder tone until going back on hook again.

In addition to not reflecting what most commercial switches
actually do, it's very common for switches to automatically
reoriginate for the user so that dial tone is provided without
the user having to depress and release the hookswitch manually.
This can increase convenience for users.

This behavior is now supported for kewlstart FXS channels.
It's supported only for kewlstart (FXOKS) mainly because the
behavior doesn't make any sense for ground start channels,
and loop start signalling doesn't provide the necessary DAHDI
event that makes this easy to implement. Likely almost everyone
is using FXOKS over FXOLS anyways since FXOLS is pretty useless
these days.

ASTERISK-30357 #close

Resolves: #224

UserNote: The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.

(cherry picked from commit c16cc93a4b)
2023-09-06 16:55:59 +00:00
Joshua C. Colp fcf7187cd6 audiohook: Unlock channel in mute if no audiohooks present.
In the case where mute was called on a channel that had no
audiohooks the code was not unlocking the channel, resulting
in a deadlock.

Resolves: #233
(cherry picked from commit 5ebe1b61bd)
2023-09-06 16:55:59 +00:00
Naveen Albert 18aa912f46 sig_analog: Allow three-way flash to time out to silence.
sig_analog allows users to flash and use the three-way dial
tone as a primitive hold function, simply by never timing
it out.

Some systems allow this dial tone to time out to silence,
so the user is not annoyed by a persistent dial tone.
This option allows the dial tone to time out normally to
silence.

ASTERISK-30004 #close
Resolves: #205

UserNote: The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.

(cherry picked from commit 5516763e57)
2023-09-06 16:55:59 +00:00
Holger Hans Peter Freyther 986c71953c res_prometheus: Do not generate broken metrics
In 8d6fdf9c3a invisible bridges were
skipped but that lead to producing metrics with no name and no help.

Keep track of the number of metrics configured and then only emit these.
Add a basic testcase that verifies that there is no '(NULL)' in the
output.

ASTERISK-30474

(cherry picked from commit a10de8bc72)
2023-09-06 16:55:59 +00:00
Sean Bright e05f51a8ae res_pjsip: Enable TLS v1.3 if present.
Fixes #221

UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.

(cherry picked from commit b5762cd54e)
2023-09-06 16:55:59 +00:00
phoneben cbcde0624b func_cut: Add example to documentation.
This adds an example to the XML documentation clarifying usage
of the CUT function to address a common misusage.

(cherry picked from commit c0b8adfe4d)
2023-09-06 16:55:59 +00:00
Sean Bright 4ed17ced6c extensions.conf.sample: Remove reference to missing context.
c3ff4648 removed the [iaxtel700] context but neglected to remove
references to it.

This commit addresses that and also removes iaxtel and freeworlddialup
references from other config files.

(cherry picked from commit df87ada8ac)
2023-09-06 16:55:59 +00:00
Sean Bright 29fbf1b2c8 func_export: Use correct function argument as variable name.
Fixes #208

(cherry picked from commit b8aee4a2c6)
2023-09-06 16:55:59 +00:00
Joshua C. Colp ef7a9e91fe app_queue: Add support for applying caller priority change immediately.
The app_queue module provides both an AMI action and a CLI command
to change the priority of a caller in a queue. Up to now this change
of priority has only been reflected to new callers into the queue.

This change adds an "immediate" option to both the AMI action and
CLI command which immediately applies the priority change respective
to the other callers already in the queue. This can allow, for example,
a caller to be placed at the head of the queue immediately if their
priority is sufficient.

Resolves: #202

UserNote: The 'queue priority caller' CLI command and
'QueueChangePriorityCaller' AMI action now have an 'immediate'
argument which allows the caller priority change to be reflected
immediately, causing the position of a caller to move within the
queue depending on the priorities of the other callers.

(cherry picked from commit 879d4ff56a)
2023-09-06 16:55:59 +00:00
George Joseph c0898cb265 .github: Fix cherry-pick reminder issues
(cherry picked from commit 12f231c717)
2023-09-06 16:55:59 +00:00
Sean Bright 22f31ecedd chan_iax2.c: Avoid crash with IAX2 switch support.
A change made in 82cebaa0 did not properly handle the case when a
channel was not provided, triggering a crash. ast_check_hangup(...)
does not protect against NULL pointers.

Fixes #180

(cherry picked from commit d26a00ef34)
2023-09-06 16:55:59 +00:00
Sean Bright db8601dd77 res_geolocation: Ensure required 'location_info' is present.
Fixes #189

(cherry picked from commit e120694711)
2023-09-06 16:55:59 +00:00
Mike Bradeen dd76dfdd57 Adds manager actions to allow move/remove/forward individual messages
in a particular mailbox folder. The forward command can be used
to copy a message within a mailbox or to another mailbox. Also adds
a VoicemailBoxSummarry, required to retrieve message ID's.

Resolves: #181

UserNote: The following manager actions have been added

VoicemailBoxSummary - Generate message list for a given mailbox

VoicemailRemove - Remove a message from a mailbox folder

VoicemailMove - Move a message from one folder to another within a mailbox

VoicemailForward - Copy a message from one folder in one mailbox
to another folder in another or the same mailbox.

(cherry picked from commit b8db3dda7e)
2023-09-06 16:55:59 +00:00
Mike Bradeen d9c4a37aaa app_voicemail: add CLI commands for message manipulation
Adds CLI commands to allow move/remove/forward individual messages
from a particular mailbox folder. The forward command can be used
to copy a message within a mailbox or to another mailbox. Also adds
a show mailbox, required to retrieve message ID's.

Resolves: #170

UserNote: The following CLI commands have been added to app_voicemail

voicemail show mailbox <mailbox> <context>
Show contents of mailbox <mailbox>@<context>

voicemail remove <mailbox> <context> <from_folder> <messageid>
Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>

voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>

voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder>
Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
mailbox <mailbox>@<context> <to_folder>

(cherry picked from commit 9b5c29d943)
2023-09-06 16:55:59 +00:00
zhengsh 37b1ceab2b res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
From the gdb information, it was found that when calling __ast_free, the size of the
allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid
is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb,
it is found to be 1.

Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid,
which is outside the protection of the rtp_instance lock. However,
ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses
rtp->themssrc_valid within the protection of the rtp_instance lock.

This can lead to the possibility that the value of rtp->themssrc_valid used in the call to
ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used
within ast_rtcp_generate_report().

Resolves: asterisk#63
(cherry picked from commit 25a766f49d)
2023-09-06 16:55:59 +00:00
George Joseph 5cf2c9f0e4 .github: Minor tweak to Asterisk Releaser
(cherry picked from commit 0f9de8a3f0)
2023-09-06 16:55:59 +00:00
George Joseph e33e6c6429 .github: Suppress cherry-pick reminder for some situations
In PROpenedOrUpdated, the cherry-pick reminder will now be
suppressed if there are already valid 'cherry-pick-to' comments
in the PR or the PR contained a 'cherry-pick-to: none' comment.

(cherry picked from commit cd7e513087)
2023-09-06 16:55:59 +00:00
Naveen Albert 967e8f6beb sig_analog: Allow immediate fake ring to be suppressed.
When immediate=yes on an FXS channel, sig_analog will
start fake audible ringback that continues until the
channel is answered. Even if it answers immediately,
the ringback is still audible for a brief moment.
This can be disruptive and unwanted behavior.

This adds an option to disable this behavior, though
the default behavior remains unchanged.

ASTERISK-30003 #close
Resolves: #118

UserNote: The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.

(cherry picked from commit d87abb2ec9)
2023-09-06 16:55:59 +00:00
Asterisk Development Team 9e2b9424f5 Update for 20.4.0 2023-07-20 13:04:41 +00:00
Asterisk Development Team 3a0d24d969 Update for 20.4.0-rc2 2023-07-13 13:58:58 +00:00
George Joseph 6cfafdeb37 app.h: Move declaration of ast_getdata_result before its first use
The ast_app_getdata() and ast_app_getdata_terminator() declarations
in app.h were changed recently to return enum ast_getdata_result
(which is how they were defined in app.c).  The existing
declaration of ast_getdata_result in app.h was about 1000 lines
after those functions however so under certain circumstances,
a "use before declaration" error was thrown by the compiler.
The declaration of the enum was therefore moved to before those
functions.

Resolves: #200
2023-07-13 05:45:06 -06:00
George Joseph 9b3abda528 doc: Remove obsolete CHANGES-staging and UPGRADE-staging 2023-07-13 05:44:51 -06:00
Asterisk Development Team c35471ad10 Update for 20.4.0-rc1 2023-07-10 11:49:53 +00:00
George Joseph 1491da61eb .github: Updates for AsteriskReleaser
(cherry picked from commit 31663fed53)
2023-07-10 11:49:48 +00:00
Mike Bradeen d180c58c36 app_voicemail: fix imap compilation errors
Fixes two compilation errors in app_voicemail_imap, one due to an unsed
variable and one due to a new variable added in the incorrect location
in _163.

Resolves: #174
(cherry picked from commit 2711fba4b9)
2023-07-10 11:49:48 +00:00
Mike Bradeen 9dbd80ab73 res_musiconhold: avoid moh state access on unlocked chan
Move channel unlock to after moh state access to avoid
potential unlocked access to state.

Resolves: #133
(cherry picked from commit 9c889911ad)
2023-07-10 11:49:48 +00:00
Mike Bradeen 714950bd7c utils: add lock timestamps for DEBUG_THREADS
Adds last locked and unlocked timestamps as well as a
counter for the number of times the lock has been
attempted (vs locked/unlocked) to debug output printed
using the DEBUG_THREADS option.

Resolves: #110
(cherry picked from commit 3acdffc17e)
2023-07-10 11:49:48 +00:00
George Joseph 244e0b69ea .github: Back out triggering PROpenedOrUpdated by label
(cherry picked from commit 4ff17f44a4)
2023-07-10 11:49:48 +00:00
George Joseph 37d2378073 .github: Move publish docs to new file CreateDocs.yml
(cherry picked from commit caf0fd357c)
2023-07-10 11:49:48 +00:00
George Joseph 46bb9e8fde rest-api: Updates for new documentation site
The new documentation site uses traditional markdown instead
of the Confluence flavored version.  This required changes in
the mustache templates and the python that generates the files.

(cherry picked from commit 80d042cb01)
2023-07-10 11:49:48 +00:00
George Joseph 64ecad8309 .github: Remove result check from PROpenUpdateGateTests
(cherry picked from commit b008fd5919)
2023-07-10 11:49:48 +00:00
George Joseph f6630916f4 .github: Fix use of 'contains'
(cherry picked from commit 44c0e1b756)
2023-07-10 11:49:48 +00:00
George Joseph 157d478b27 .github: Add recheck label test to additional jobs
(cherry picked from commit 91503078ff)
2023-07-10 11:49:48 +00:00
George Joseph 8a152fb4aa .github: Fix recheck label typos
(cherry picked from commit de40ea0f79)
2023-07-10 11:49:48 +00:00
George Joseph 37dce12781 .github: Fix recheck label manipulation
(cherry picked from commit 07a3784745)
2023-07-10 11:49:48 +00:00
George Joseph fdfe61bf4d .github: Allow PR submit checks to be re-run by label
(cherry picked from commit 426b9c568c)
2023-07-10 11:49:48 +00:00
Olaf Titz f7f2872b3f app_voicemail_imap: Fix message count when IMAP server is unavailable
Some callers of __messagecount did not correctly handle error return,
instead returning a -1 message count.
This caused a notification with "Messages-Waiting: yes" and
"Voice-Message: -1/0 (0/0)" if the IMAP server was unavailable.

Fixes: #64
(cherry picked from commit af2ced484e)
2023-07-10 11:49:48 +00:00
Sean Bright 81b490ab0a res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
Resolves: #116
(cherry picked from commit 4a1edb9a8c)
2023-07-10 11:49:48 +00:00
Ben Ford ff4f3c1187 res_pjsip_session: Added new function calls to avoid ABI issues.
Added two new functions (ast_sip_session_get_dialog and
ast_sip_session_get_pjsip_inv_state) that retrieve the dialog and the
pjsip_inv_state respectively from the pjsip_inv_session on the
ast_sip_session struct. This is due to pjproject adding a new field to
the pjsip_inv_session struct that caused crashes when trying to access
fields that were no longer where they were expected to be if a module
was compiled against a different version of pjproject.

Resolves: #145
(cherry picked from commit ad120e5d0b)
2023-07-10 11:49:48 +00:00
Nathan Bruning f0df2eba9d app_queue: Add force_longest_waiting_caller option.
This adds an option 'force_longest_waiting_caller' which changes the
global behavior of the queue engine to prevent queue callers from
'jumping ahead' when an agent is in multiple queues.

Resolves: #108

Also closes old asterisk issues:
- ASTERISK-17732
- ASTERISK-17570

Change-Id: I0f84e27903fefbe2018d0afa2d67b23aa0b321ce
(cherry picked from commit 6084bbfe0e)
2023-07-10 11:49:48 +00:00
Sean Bright ff60aae1c8 pjsip_transport_events.c: Use %zu printf specifier for size_t.
Partially resolves #143.

(cherry picked from commit b7eae29fb9)
2023-07-10 11:49:48 +00:00
Sean Bright 99622c35e0 res_crypto.c: Gracefully handle potential key filename truncation.
Partially resolves #143.

(cherry picked from commit 97b901a29b)
2023-07-10 11:49:48 +00:00
Sean Bright 5497b1abe3 configure: Remove obsolete and deprecated constructs.
These were uncovered when trying to run `bootstrap.sh` with Autoconf
2.71:

* AC_CONFIG_HEADER() is deprecated in favor of AC_CONFIG_HEADERS().
* AC_HEADER_TIME is obsolete.
* $as_echo is deprecated in favor of AS_ECHO() which requires an update
  to ax_pthread.m4.

Note that the generated artifacts in this commit are from Autoconf 2.69.

Resolves #139

(cherry picked from commit 21d3c4ae58)
2023-07-10 11:49:48 +00:00
zhou_jiajian 3ad8ab6d52 res_fax_spandsp.c: Clean up a spaces/tabs issue
(cherry picked from commit f7e1f1fbbd)
2023-07-10 11:49:48 +00:00
Sean Bright f971db6685 ast-db-manage: Synchronize revisions between comments and code.
In a handful of migrations, the comment header that indicates the
current and previous revisions has drifted from the identifiers
revision and down_revision variables. This updates the comment headers
to match the code.

(cherry picked from commit ea63148b23)
2023-07-10 11:49:48 +00:00
George Joseph 8e6f15d185 test_statis_endpoints: Fix channel_messages test again
(cherry picked from commit 25bc5544d8)
2023-07-10 11:49:48 +00:00
Sean Bright 0503e2b612 res_crypto.c: Avoid using the non-portable ALLPERMS macro.
ALLPERMS is not POSIX and it's trivial enough to not jump through
autoconf hoops to check for it.

Fixes #149.

(cherry picked from commit ebc007322b)
2023-07-10 11:49:48 +00:00
Jaco Kroon dafa9917ff tcptls: when disabling a server port, we should set the accept_fd to -1.
If we don't set this to -1 if the structure can be potentially re-used
later then it's possible that we'll issue a close() on an unrelated file
descriptor, breaking asterisk in other interesting ways.

I believe this to be an unlikely scenario, but it costs nothing to be
safe.

Signed-off-by: Jaco Kroon <jaco@uls.co.za>
(cherry picked from commit 4a637d6d11)
2023-07-10 11:49:48 +00:00
Jiajian Zhou a6a5498b0d AMI: Add parking position parameter to Park action
Add a parking space extension parameter (ParkingSpace) to the Park action.
Park action will attempt to park the call to that extension.
If the extension is already in use, then execution will continue at the next priority.

UserNote: New ParkingSpace parameter has been added to AMI action Park.
(cherry picked from commit 8e3f9dcc7b)
2023-07-10 11:49:48 +00:00
George Joseph b7775ae4e1 test_stasis_endpoints.c: Make channel_messages more stable
The channel_messages test was assuming that stasis would return
messages in a specific order.  This is an incorrect assumption as
message ordering was never guaranteed.  This was causing the test
to fail occasionally.  We now test all the messages for the
required message types instead of testing one by one.

Resolves: #158
(cherry picked from commit 83781c23b9)
2023-07-10 11:49:48 +00:00
George Joseph ee09bbbb5f build: Fix a few gcc 13 issues
* gcc 13 is now catching when a function is declared as returning
  an enum but defined as returning an int or vice versa.  Fixed
  a few in app.h, loader.c, stasis_message.c.

* gcc 13 is also now (incorrectly) complaining of dangling pointers
  when assigning a pointer to a local char array to a char *. Had
  to change that to an ast_alloca.

Resolves: #155
(cherry picked from commit 6c8b23a688)
2023-07-10 11:49:48 +00:00
George Joseph 40ff67d636 .github: Rework for merge approval
(cherry picked from commit b3c2a9cd44)
2023-07-10 11:49:48 +00:00
Sean Bright 79d271fa7f ast-db-manage: Fix alembic branching error caused by #122.
Fixes #147.

(cherry picked from commit fe15631d38)
2023-07-10 11:49:48 +00:00
alex2grad 319da11fae app_followme: fix issue with enable_callee_prompt=no (#88)
* app_followme: fix issue with enable_callee_prompt=no

If the FollowMe option 'enable_callee_prompt' is set to 'no' then Asterisk
incorrectly sets a winner channel to the channel from which any control frame was read.

This fix sets the winner channel only to the answered channel.

Resolves: #87

ASTERISK-30326

(cherry picked from commit a8ea16cdf8)
2023-07-10 11:49:48 +00:00
Sean Bright 228244a96e sounds: Update download URL to use HTTPS.
Related to #136

(cherry picked from commit ada3dc2adb)
2023-07-10 11:49:48 +00:00
Miguel Angel Nubla 25de041d5e configure: Makefile downloader enable follow redirects.
If curl is used for building, any download such as a sounds package
will fail to follow HTTP redirects and will download wrong data.

Resolves: #136
(cherry picked from commit 95d339ac7a)
2023-07-10 11:49:47 +00:00
Naveen Albert 8cdf711531 res_musiconhold: Add option to loop last file.
Adds the loop_last option to res_musiconhold,
which allows the last audio file in the directory
to be looped perpetually once reached, rather than
circling back to the beginning again.

Resolves: #122
ASTERISK-30462

UserNote: The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.

(cherry picked from commit edf488c76e)
2023-07-10 11:49:47 +00:00
Naveen Albert 457a72304d chan_dahdi: Fix Caller ID presentation for FXO ports.
Currently, the presentation for incoming channels is
always available, because it is never actually set,
meaning the channel presentation can be nonsensical.
If the presentation from the incoming Caller ID spill
is private or unavailable, we now update the channel
presentation to reflect this.

Resolves: #120
ASTERISK-30333
ASTERISK-21741

(cherry picked from commit 86a11d5b19)
2023-07-10 11:49:47 +00:00
Ben Ford 8da7b4606b AMI: Add CoreShowChannelMap action.
Adds a new AMI action (CoreShowChannelMap) that takes in a channel name
and provides a list of all channels that are connected to that channel,
following local channel connections as well.

Resolves: #104

UserNote: New AMI action CoreShowChannelMap has been added.
(cherry picked from commit e72b277828)
2023-07-10 11:49:47 +00:00
Naveen Albert f40e50ce19 sig_analog: Add fuller Caller ID support.
A previous change, ASTERISK_29991, made it possible
to send additional Caller ID parameters that were
not previously supported.

This change adds support for analog DAHDI channels
to now be able to receive these parameters for
on-hook Caller ID, in order to enhance the usability
of CPE that support these parameters.

Resolves: #94
ASTERISK-30331

UserNote: Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.

(cherry picked from commit 87f44dc0f9)
2023-07-10 11:49:47 +00:00
Joe Searle 5018c09d99 res_stasis.c: Add new type 'sdp_label' for bridge creation.
Add new type 'sdp_label' when creating a bridge using the ARI. This will
add labels to the SDP for each stream, the label is set to the
corresponding channel id.

Resolves: #91

UserNote: When creating a bridge using the ARI the 'type' argument now
accepts a new value 'sdp_label' which will configure the bridge to add
labels for each stream in the SDP with the corresponding channel id.

(cherry picked from commit ec64828c8f)
2023-07-10 11:49:47 +00:00
Niklas Larsson 1589efa5dd app_queue: Preserve reason for realtime queues
When Asterisk is restarted it does not preserve paused reason for
members of realtime queues. This was fixed for non-realtime queues in
ASTERISK_25732

Resolves: #66

UpgradeNote: Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.

UserNote: Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.

(cherry picked from commit df774619fb)
2023-07-10 11:49:47 +00:00
George Joseph b68bf5683f .github: Fix issues with cherry-pick-reminder
(cherry picked from commit a83f35c58e)
2023-07-10 11:49:47 +00:00
Mike Bradeen 26401ddcc0 indications: logging changes
Increase verbosity to indicate failure due to missing country
and to specify default on CLI dump

Resolves: #89
(cherry picked from commit f19a6cf11e)
2023-07-10 11:49:47 +00:00
George Joseph da66f402c9 .github Ignore error when adding reviewrs to PR
(cherry picked from commit 7bbeda3908)
2023-07-10 11:49:47 +00:00
George Joseph 75c13b2c95 .github: Update field descriptions for AsteriskReleaser
(cherry picked from commit 2a75114b6d)
2023-07-10 11:49:47 +00:00
Naveen Albert ba1c363d3e callerid: Allow specifying timezone for date/time.
The Caller ID generation routine currently is hardcoded
to always use the system time zone. This makes it possible
to optionally specify any TZ-format time zone.

Resolves: #98
ASTERISK-30330

(cherry picked from commit d3227a614a)
2023-07-10 11:49:47 +00:00
Naveen Albert c8b7570cad logrotate: Fix duplicate log entries.
The Asterisk logrotate script contains explicit
references to files with the .log extension,
which are also included when *log is expanded.
This causes issues with newer versions of logrotate.
This fixes this by ensuring that a log file cannot
be referenced multiple times after expansion occurs.

Resolves: #96
ASTERISK-30442
Reported by: EN Barnett
Tested by: EN Barnett

(cherry picked from commit 22c9d52289)
2023-07-10 11:49:47 +00:00
Maximilian Fridrich 173c433274 chan_pjsip: Allow topology/session refreshes in early media state
With this change, session modifications in the early media state are
possible if the SDP was sent reliably and confirmed by a PRACK. For
details, see RFC 6337, escpecially section 3.2.

Resolves: #73
(cherry picked from commit 1a7dada804)
2023-07-10 11:49:47 +00:00
Naveen Albert c5c89ed377 chan_dahdi: Fix broken hidecallerid setting.
The hidecallerid setting in chan_dahdi.conf currently
is broken for a couple reasons.

First, the actual code in sig_analog to "allow" or "block"
Caller ID depending on this setting improperly used
ast_set_callerid instead of updating the presentation.
This issue was mostly fixed in ASTERISK_29991, and that
fix is carried forward to this code as well.

Secondly, the hidecallerid setting is set on the DAHDI
pvt but not carried forward to the analog pvt properly.
This is because the chan_dahdi config loading code improperly
set permhidecallerid to permhidecallerid from the config file,
even though hidecallerid is what is actually set from the config
file. (This is done correctly for call waiting, a few lines above.)
This is fixed to read the proper value.

Thirdly, in sig_analog, hidecallerid is set to permhidecallerid
only on hangup. This can lead to potential security vulnerabilities
as an allowed Caller ID from an initial call can "leak" into subsequent
calls if no hangup occurs between them. This is fixed by setting
hidecallerid to permcallerid when calls begin, rather than when they end.
This also means we don't need to also set hidecallerid in chan_dahdi.c
when copying from the config, as we would have to otherwise.

Fourthly, sig_analog currently only allows dialing *67 or *82 if
that would actually toggle the presentation. A comment is added
clarifying that this behavior is okay.

Finally, a couple log messages are updated to be more accurate.

Resolves: #100
ASTERISK-30349 #close

(cherry picked from commit 22b599d86f)
2023-07-10 11:49:47 +00:00
George Joseph 0d214b6611 .github: Change title of AsteriskReleaser job
(cherry picked from commit 15439d08bd)
2023-07-10 11:49:47 +00:00
Naveen Albert ca51a25056 asterisk.c: Fix option warning for remote console.
Commit 09e989f972
categorized the T option as not being compatible
with remote consoles, but they do affect verbose
messages with remote console. This fixes this.

Resolves: #102
(cherry picked from commit cfae64a70d)
2023-07-10 11:49:47 +00:00
George Joseph cd2eafe588 .github: Don't add cherry-pick reminder if it's already present
(cherry picked from commit 7193e95676)
2023-07-10 11:49:47 +00:00
George Joseph 955c2e06ff .github: Fix quoting in PROpenedOrUpdated
(cherry picked from commit af0535f302)
2023-07-10 11:49:47 +00:00
George Joseph 5e37887d82 .github: Add cherry-pick reminder to new PRs
(cherry picked from commit e72b35e7fb)
2023-07-10 11:49:47 +00:00
Jaco Kroon dda4dd513f configure: fix test code to match gethostbyname_r prototype.
This enables the test to work with CC=clang.

Without this the test for 6 args would fail with:

utils.c:99:12: error: static declaration of 'gethostbyname_r' follows non-static declaration
static int gethostbyname_r (const char *name, struct hostent *ret, char *buf,
           ^
/usr/include/netdb.h:177:12: note: previous declaration is here
extern int gethostbyname_r (const char *__restrict __name,
           ^

Fixing the expected return type to int sorts this out.

Signed-off-by: Jaco Kroon <jaco@uls.co.za>
(cherry picked from commit f178bf78b7)
2023-07-10 11:49:47 +00:00
Sean Bright bbf47fefab res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
The functionality we are interested in is present only in pjsip 2.13
and newer.

Resolves: #45
(cherry picked from commit 2d18fb6e9d)
2023-07-10 11:49:47 +00:00
zhengsh 2d1e5f81ee res_sorcery_memory_cache.c: Fix memory leak
Replace the original call to ast_strdup with a call to ast_strdupa to fix the leak issue.

Resolves: #55
ASTERISK-30429

(cherry picked from commit 8abb294063)
2023-07-10 11:49:47 +00:00
Sean Bright 0e4a9d7251 xml.c: Process XML Inclusions recursively.
If processing an XInclude results in new <xi:include> elements, we
need to run XInclude processing again. This continues until no
replacement occurs or an error is encountered.

There is a separate issue with dynamic strings (ast_str) that will be
addressed separately.

Resolves: #65
(cherry picked from commit 19973b73fd)
2023-07-10 11:49:47 +00:00
Joshua C. Colp 6812357bef .github: Tweak improvement issue type language.
(cherry picked from commit f0e2d45089)
2023-07-10 11:49:47 +00:00
Gitea d35a0d3d4e .github: Tweak new feature language, and move feature requests elsewhere.
(cherry picked from commit c0e32d8245)
2023-07-10 11:49:47 +00:00
Joshua C. Colp 31c11bec74 .github: Fix staleness check to only run on certain labels.
(cherry picked from commit 851ec44714)
2023-07-10 11:49:47 +00:00
Asterisk Development Team 6dc1877b2d Update for 20.3.1 2023-07-07 20:09:14 +00:00
Sean Bright 9ca24c9c2b apply_patches: Use globbing instead of file/sort.
This accomplishes the same thing as a `find ... | sort` but with the
added benefit of clarity and avoiding a call to a subshell.

Additionally drop the -s option from call to patch as it is not POSIX.
2023-07-07 10:02:18 -06:00
George Joseph 90069a94b0 apply_patches: Sort patch list before applying
The apply_patches script wasn't sorting the list of patches in
the "patches" directory before applying them. This left the list
in an indeterminate order. In most cases, the list is actually
sorted but rarely, they can be out of order and cause dependent
patches to fail to apply.

We now sort the list but the "sort" program wasn't in the
configure scripts so we needed to add that and regenerate
the scripts as well.

Resolves: #193
2023-07-07 10:02:02 -06:00
Stanislav Abramenkov 0fa022a39c pjsip: Upgrade bundled version to pjproject 2.13.1 2023-07-07 10:01:51 -06:00
Asterisk Development Team 282045db34 Update for 20.3.0 2023-05-23 14:31:56 +00:00
George Joseph a004574465 Set up new ChangeLogs directory 2023-05-09 08:34:50 -06:00
Asterisk Development Team 2e1842229a Update for 20.3.0-rc1 2023-05-08 18:13:41 +00:00
George Joseph 3fe0d8088f .github: Add AsteriskReleaser
(cherry picked from commit 9d99f6aaf1)
2023-05-08 18:13:36 +00:00
Henning Westerholt ac747ce562 chan_pjsip: also return all codecs on empty re-INVITE for late offers
We should also return all codecs on an re-INVITE without SDP for a
call that used late offer (e.g. no SDP in the initial INVITE, SDP
in the ACK). Bugfix for feature introduced in ASTERISK-30193
(https://issues.asterisk.org/jira/browse/ASTERISK-30193)

Migration from previous gerrit change that was not merged.

(cherry picked from commit 37a41a2a20)
2023-05-08 18:13:36 +00:00
Mike Bradeen 21ee5f705a cel: add local optimization begin event
The current AST_CEL_LOCAL_OPTIMIZE event is and has been
triggered on a local optimization end to serve as a flag
indicating the event occurred.  This change adds a second
AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail.

Resolves: #52

UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.

UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.

(cherry picked from commit 8d3d8fcdcd)
2023-05-08 18:13:36 +00:00
Sean Bright 7677e78b4f core: Cleanup gerrit and JIRA references. (#57)
* 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
(cherry picked from commit 5c6d5ea38f)
2023-05-08 18:13:36 +00:00
George Joseph 9a1ac9142f .github: Fix CherryPickTest to only run when it should
Fixed CherryPickTest so it triggers only on the
"cherry-pick-test" label instead of all labels.

(cherry picked from commit 9097fc7515)
2023-05-08 18:13:36 +00:00
George Joseph 272fbca4d0 .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
(cherry picked from commit fbde0a71f0)
2023-05-08 18:13:36 +00:00
George Joseph e3b9f66678 .github: Remove separate set labels step from new PR
(cherry picked from commit aced9046ca)
2023-05-08 18:13:36 +00:00
George Joseph 6aa490d373 .github: Refactor CP progress and add new PR test progress
(cherry picked from commit b7eeef6314)
2023-05-08 18:13:36 +00:00
Maximilian Fridrich 9cfda39d36 res_pjsip: mediasec: Add Security-Client headers after 401
When using mediasec, requests sent after a 401 must still contain the
Security-Client header according to
draft-dawes-sipcore-mediasec-parameter.

Resolves: #48
(cherry picked from commit f3cc1e7fbd)
2023-05-08 18:13:36 +00:00
George Joseph ea564b640d .github: Add cherry-pick test progress labels
(cherry picked from commit 3707140472)
2023-05-08 18:13:36 +00:00
Joshua C. Colp 42c243866e LICENSE: Update link to trademark policy.
Resolves: #43
(cherry picked from commit a614604f32)
2023-05-08 18:13:36 +00:00
Naveen Albert e6b84eca75 chan_dahdi: Add dialmode option for FXS lines.
Currently, both pulse and tone dialing are always enabled
on all FXS lines, with no way of disabling one or the other.

In some circumstances, it is desirable or necessary to
disable one of these, and this behavior can be problematic.

A new "dialmode" option is added which allows setting the
methods to support on a per channel basis for FXS (FXO
signalled lines). The four options are "both", "pulse",
"dtmf"/"tone", and "none".

Additionally, integration with the CHANNEL function is
added so that this setting can be updated for a channel
during a call.

Resolves: #35
ASTERISK-29992

UserNote: A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.

Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.

(cherry picked from commit 82d7bb49dd)
2023-05-08 18:13:36 +00:00
George Joseph 81310a031c .github: Update issue templates
(cherry picked from commit ef644c3e93)
2023-05-08 18:13:35 +00:00
George Joseph 5cb6d8a453 .github: Remove unnecessary parameter in CherryPickTest
(cherry picked from commit 313cd9dd84)
2023-05-08 18:13:35 +00:00
George Joseph cc36ae8de5 Initial GitHub PRs
(cherry picked from commit 987315a5fa)
2023-05-08 18:13:35 +00:00
George Joseph f994463aee Initial GitHub Issue Templates
(cherry picked from commit f8dc014819)
2023-05-08 18:13:35 +00:00
Joshua C. Colp 6a347f413d pbx_dundi: Fix PJSIP endpoint configuration check.
ASTERISK-28233

Change-Id: I0f11c096b307a6178e22ca49d9c756343f0e1fdc
(cherry picked from commit 6c53e5e870)
2023-05-08 18:13:35 +00:00
Joshua Colp 9c5f5358d7 Revert "app_queue: periodic announcement configurable start time."
This reverts commit 3fd0b65bae.

Reason for revert: Causes segmentation fault.

Change-Id: Ic189c6f7872943a5500d3e71142f0c09d54fcc31
(cherry picked from commit de15852ef0)
2023-05-08 18:13:35 +00:00
Naveen Albert 59c9e4531e res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
The current STIR/SHAKEN signing process is inconsistent with the
RFCs in a couple ways that can cause interoperability issues.

RFC8225 specifies that the keys must be ordered lexicographically, but
currently the fields are simply ordered according to the order
in which they were added to the JSON object, which is not
compliant with the RFC and can cause issues with some carriers.

To fix this, we now leverage libjansson's ability to dump a JSON
object sorted by key value, yielding the correct field ordering.

Additionally, telephone numbers must have any leading + prefix removed
and must not contain characters outside of 0-9, *, and # in order
to comply with the RFCs. Numbers are now properly formatted as such.

ASTERISK-30407 #close

Change-Id: Iab76d39447c4b8cf133de85657dba02fda07f9a2
(cherry picked from commit 66a7cff57e)
2023-05-08 18:13:35 +00:00
Naveen Albert 99d17db963 pbx_dundi: Add PJSIP support.
Adds PJSIP as a supported technology to DUNDi.

To facilitate this, we now allow an endpoint to be specified
for outgoing PJSIP calls. We also allow users to force a specific
channel technology for outgoing SIP-protocol calls.

ASTERISK-28109 #close
ASTERISK-28233 #close

Change-Id: I2e28e5a5d007bd49e3df113ad567b011103899bf
(cherry picked from commit b33f92cbb5)
2023-05-08 18:13:35 +00:00
The_Blode 81a912f3ce install_prereq: Add Linux Mint support.
ASTERISK-30359 #close

Change-Id: I9c140c7f12ca7dafe65c317f2a26122cf2c72556
(cherry picked from commit de9aaf7e97)
2023-05-08 18:13:35 +00:00
Henning Westerholt 51509309ca chan_pjsip: fix music on hold continues after INVITE with replaces
In a three party scenario with INVITE with replaces, we need to
unhold the call, otherwise one party continues to get music on
hold, and the call is not properly bridged between them.

ASTERISK-30428

Change-Id: I5675df11e739be5226b328f8828d4b8d81fbefb4
(cherry picked from commit 1c5720b802)
2023-05-08 18:13:35 +00:00
Naveen Albert c9aa3b0169 voicemail.conf: Fix incorrect comment about #include.
A comment at the top of voicemail.conf says that #include
cannot be used in voicemail.conf because this breaks
the ability for app_voicemail to auto-update passwords.
This is factually incorrect, since Asterisk has no problem
updating files that are #include'd in the main configuration
file, and this does work in voicemail.conf as well.

ASTERISK-30479 #close

Change-Id: I3bf7d275849ab83f55f7fb6702a75a3077ee1df3
(cherry picked from commit d0f39250ee)
2023-05-08 18:13:35 +00:00
Naveen Albert 5e7d21e5a5 app_queue: Fix minor xmldoc duplication and vagueness.
The F option in the xmldocs for the Queue application
was erroneously duplicated, causing it to display
twice on the wiki. The two sections are now merged into one.

Additionally, the description for the d option was quite
vague. Some more details are added to provide context
as to what this actually does.

ASTERISK-30486 #close

Change-Id: I6706cea708b5cc781f59f8652c2cb377e55aed7e
(cherry picked from commit bad5bda08c)
2023-05-08 18:13:35 +00:00
George Joseph d0ce6d4ee5 test.c: Fix counting of tests and add 2 new tests
The unit test XML output was counting all registered tests as "run"
even when only a subset were actually requested to be run and
the "failures" attribute was missing.

* The "tests" attribute of the "testsuite" element in the
  output XML now reflects only the tests actually requested
  to be executed instead of all the tests registered.

* The "failures" attribute was added to the "testsuite"
  element.

Also added 2 new unit tests that just pass and fail to be
used for CI testing.

Change-Id: Ia137814b5aeb0e1a44c75034bd3615c26021da69
(cherry picked from commit a0fd95ef52)
2023-05-08 18:13:35 +00:00
Jaco Kroon 3a158c07a1 res_calendar: output busy state as part of show calendar.
Change-Id: I894e4ecc3e93db4ff7783d46266ba3c5e6ccda10
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
(cherry picked from commit 019dc51139)
2023-05-08 18:13:35 +00:00
Sean Bright 255565e240 loader.c: Minor module key check simplification.
Change-Id: I65aefd4434a783096165c179b5f94f2e4810dffe
(cherry picked from commit 6dab013e49)
2023-05-08 18:13:35 +00:00
Sean Bright 6ff3258c17 ael: Regenerate lexers and parsers.
Various changes to ensure that the lexers and parsers can be correctly
generated when REBUILD_PARSERS is enabled.

Some notes:

* Because of the version of flex we are using to generate the lexers
  (2.5.35) some post-processing in the Makefile is still required.

* The generated lexers do not contain the problematic C99 check that
  was being replaced by the call to sed in the respective Makefiles so
  it was removed.

* Since these files are generated, they will include trailing
  whitespace in some places. This does not need to be corrected.

Change-Id: Ibbd343606fcf5c0d285b1599e6e8e59f514f2e4e
(cherry picked from commit 34ff836db5)
2023-05-08 18:13:35 +00:00
Mike Bradeen 2ac8388c6d bridge_builtin_features: add beep via touch variable
Add periodic beep option to one-touch recording by setting
the touch variable TOUCH_MONITOR_BEEP or
TOUCH_MIXMONITOR_BEEP to the desired interval in seconds.

If the interval is less than 5 seconds, a minimum of 5
seconds will be imposed.  If the interval is set to an
invalid value, it will default to 15 seconds.

A new test event PERIODIC_HOOK_ENABLED was added to the
func_periodic_hook hook_on function to indicate when
a hook is started.  This is so we can test that the touch
variable starts the hook as expected.

ASTERISK-30446

Change-Id: I800e494a789ba7a930bbdcd717e89d86040d6661
(cherry picked from commit ffe346b2de)
2023-05-08 18:13:35 +00:00
Mike Bradeen e00eaa74e3 res_mixmonitor: MixMonitorMute by MixMonitor ID
While it is possible to create multiple mixmonitor instances
on a channel, it was not previously possible to mute individual
instances.

This change includes the ability to specify the MixMonitorID
when calling the manager action: MixMonitorMute.  This will
allow an individual MixMonitor instance to be muted via id.
This id can be stored as a channel variable using the 'i'
MixMonitor option.

As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor spy-type audiohooks on the channel.
This is done via the new audiohook function:
ast_audiohook_set_mute_all.

ASTERISK-30464

Change-Id: Ibba8c7e750577aa1595a24b23316ef445245be98
(cherry picked from commit fa635a872e)
2023-05-08 18:13:35 +00:00
Mike Bradeen 3a4fd2fa42 format_sln: add .slin as supported file extension
Adds '.slin' to existing supported file extensions:
.sln and .raw

ASTERISK-30465

Change-Id: Ice848addc03a64c8404b87cb5d3b13399c57e496
(cherry picked from commit 8d2ffc8aa5)
2023-05-08 18:13:35 +00:00
Sean Bright b376d16d81 res_agi: RECORD FILE plays 2 beeps.
Sending the "RECORD FILE" command without the optional
`offset_samples` argument can result in two beeps playing on the
channel.

This bug has been present since Asterisk 0.3.0 (2003-02-06).

ASTERISK-30457 #close

Change-Id: I95e88aa59378784d7f0eb648843f090e6723b787
(cherry picked from commit a3ec3efa02)
2023-05-08 18:13:35 +00:00
Naveen Albert e104bd0852 func_json: Fix JSON parsing issues.
Fix issue with returning empty instead of dumping
the JSON string when recursing.

Also adds a unit test to capture this fix.

ASTERISK-30441 #close

Change-Id: If0bde9f3fe84f7af485e0838205cc21e0f752a85
(cherry picked from commit f9fd76677f)
2023-05-08 18:13:35 +00:00
Naveen Albert 3dcf6ddde5 app_senddtmf: Add SendFlash AMI action.
Adds an AMI action to send a flash event
on a channel.

ASTERISK-30440 #close

Change-Id: I4707aeecb3cd8f3e63fd0c3fe009798943c369c9
(cherry picked from commit 3556ca239a)
2023-05-08 18:13:35 +00:00
Naveen Albert 0dd0bc3e65 app_dial: Fix DTMF not relayed to caller on unanswered calls.
DTMF frames are not handled in app_dial when sent towards the
caller. This means that if DTMF is sent to the calling party
and the call has not yet been answered, the DTMF is not audible.
This is now fixed by relaying DTMF frames if only a single
destination is being dialed.

ASTERISK-29516 #close

Change-Id: Iafd7430ac2915126d42dc48f0b73b262452ee027
(cherry picked from commit 090ec448cf)
2023-05-08 18:13:35 +00:00
Fabrice Fontaine bdd2bac055 configure: fix detection of re-entrant resolver functions
uClibc does not provide res_nsearch:
asterisk-16.0.0/main/dns.c:506: undefined reference to `res_nsearch'

Patch coded by Yann E. MORIN:
http://lists.busybox.net/pipermail/buildroot/2018-October/232630.html

ASTERISK-21795 #close

Signed-off-by: Bernd Kuhls <bernd.kuhls@t-online.de>
[Retrieved from:
https: //git.buildroot.net/buildroot/tree/package/asterisk/0005-configure-fix-detection-of-re-entrant-resolver-funct.patch]
Signed-off-by: Fabrice Fontaine <fontaine.fabrice@gmail.com>
Change-Id: I79296f19e28ec764bbd1e991bf11c416d0b10563
(cherry picked from commit cb0220dec2)
2023-05-08 18:13:35 +00:00
Mike Bradeen 5a85b372af cli: increase channel column width
For 'core show channels', the Channel name field is increased
to 64 characters and the Location name field is increased to
32 characters.

For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.

ASTERISK-30455

Change-Id: Ibec3742ce360ffc93bc56e9984c2a21dabc4d5e1
(cherry picked from commit 405211eff7)
2023-05-08 18:13:35 +00:00
Jaco Kroon d6e733d4bc app_queue: periodic announcement configurable start time.
This newly introduced periodic-announce-startdelay makes it possible to
configure the initial start delay of the first periodic announcement
after which periodic-announce-frequency takes over.

ASTERISK-30437 #close
Change-Id: Ia79984b6377ef78f167ad9ea2ac084bec29955d0
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
(cherry picked from commit 3fd0b65bae)
2023-05-08 18:13:35 +00:00
George Joseph bd1f8b89da make_version: Strip svn stuff and suppress ref HEAD errors
* All of the code that used subversion has been removed.

* When Asterisk is checked out from a tag or commit instead
  of one of the regular branches, git would emit messages like
  "fatal: ref HEAD is not a symbolic ref" which weren't fatal
  at all.  Those are now suppressed.

Change-Id: I2a11bc9ebbaf6dfa50f53516ede50a6bac65ca3c
(cherry picked from commit bbec5d1a99)
2023-05-08 18:13:35 +00:00
Holger Hans Peter Freyther 723c9093ad res_http_media_cache: Introduce options and customize
Make the existing CURL parameters configurable and allow
to specify the usable protocols, proxy and DNS timeout.

ASTERISK-30340

Change-Id: I2eb02ef44190e026716720419bcbdbcc8125777b
(cherry picked from commit 8f088aa0f7)
2023-05-08 18:13:35 +00:00
Fabrice Fontaine b17177c419 main/iostream.c: fix build with libressl
Fix the following build failure with libressl by using SSL_is_server
which is available since version 2.7.0 and
d7ec516916:

iostream.c: In function 'ast_iostream_close':
iostream.c:559:41: error: invalid use of incomplete typedef 'SSL' {aka 'struct ssl_st'}
  559 |                         if (!stream->ssl->server) {
      |                                         ^~

ASTERISK-30107 #close

Fixes: - http://autobuild.buildroot.org/results/ce4d62d00bb77ba5b303cacf6be7e350581a62f9
Change-Id: Iea7f34970297f2fb50285d73462d0174ba7e9587
(cherry picked from commit 030b7b9009)
2023-05-08 18:13:35 +00:00
Sean Bright 61c8f9b225 contrib: rc.archlinux.asterisk uses invalid redirect.
`rc.archlinux.asterisk`, which explicitly requests bash in its
shebang, uses the following command syntax:

  ${DAEMON} -rx "core stop now" > /dev/null 2&>1

The intent of which is to execute:

  ${DAEMON} -rx "core stop now"

While sending both stdout and stderr to `/dev/null`. Unfortunately,
because the `&` is in the wrong place, bash is interpreting the `2` as
just an additional argument to the `$DAEMON` command and not as a file
descriptor and proceeds to use the bashism `&>` to send stderr and
stdout to a file named `1`.

So we clean it up and just use bash's shortcut syntax.

Issue raised and a fix suggested (but not used) by peutch on GitHub¹.

ASTERISK-30449 #close

1. https://github.com/asterisk/asterisk/pull/31

Change-Id: Ie279bf4efb4d95cbf507313483d316e977303d19
(cherry picked from commit 46bdd5e3be)
2023-05-08 18:13:35 +00:00
Asterisk Development Team 1d64cb99d2 Update for 20.2.1 2023-04-03 10:49:49 -05:00
Mike Bradeen f8dfbaf225 res_pjsip_pubsub: subscription cleanup changes
There are two main parts of the change associated with this
commit. These are driven by the change in call order of
pubsub_on_rx_refresh and pubsub_on_evsub_state by pjproject
when an in-dialog SUBSCRIBE is received.

First, the previous behavior was for pjproject to call
pubsub_on_rx_refresh before calling pubsub_on_evsub_state
when an in-dialog SUBSCRIBE was received that changes the
subscription state.

If that change was a termination due to a re-SUBSCRIBE with
an expires of 0, we used to use the call to pubsub_on_rx_refresh
to set the substate of the evsub to TERMINATE_PENDING before
pjproject could call pubsub_on_evsub_state.

This substate let pubsub_on_evsub_state know that the
subscription TERMINATED event could be ignored as there was
still a subsequent NOTIFY that needed to be generated and
another call to pubsub_on_evsub_state to come with it.

That NOTIFY was sent via serialized_pubsub_on_refresh_timeout
which would see the TERMINATE_PENDING state and transition it
to TERMINATE_IN_PROGRESS before triggering another call to
pubsub_on_evsub_state (which now would clean up the evsub.)

The new pjproject behavior is to call pubsub_on_evsub_state
before pubsub_on_rx_refresh. This means we no longer can set
the state to TERMINATE_PENDING to tell pubsub_on_evsub_state
that it can ignore the first TERMINATED event.

To handle this, we now look directly at the event type,
method type and the expires value to determine whether we
want to ignore the event or use it to trigger the evsub
cleanup.

Second, pjproject now expects the NOTIFY to actually be sent
during pubsub_on_rx_refresh and avoids the protocol violation
inherent in sending a NOTIFY before the SUBSCRIBE is
acknowledged by caching the sent NOTIFY then sending it
after responding to the SUBSCRIBE.

This requires we send the NOTIFY using the non-serialized
pubsub_on_refresh_timeout directly and let pjproject handle
the protocol violation.

ASTERISK-30469

Change-Id: I05c1d91a44fe28244ae93faa4a2268a3332b5fd7
2023-04-03 09:43:55 -05:00
Sean Bright 6e50550d28 Revert "pbx_ael: Global variables are not expanded."
This reverts commit 56051d1ac5.

Reason for revert: Behavior change that breaks existing dialplan.

ASTERISK-30472 #close

Change-Id: I83bed3b800d36228a04ded0a6164b795f7f16bd6
2023-03-20 06:50:49 -06:00
Asterisk Development Team e7f0440032 Update for 20.2.0 2023-03-09 12:17:03 -05:00
Asterisk Development Team 460d13e916 Update for 20.2.0-rc1 2023-03-02 11:45:57 -05:00
Asterisk Development Team 93813c9dca Update CHANGES and UPGRADE.txt for 20.2.0 2023-03-02 11:37:42 -05:00
George Joseph ceda5a9859 res_pjsip: Replace invalid UTF-8 sequences in callerid name
* Added a new function ast_utf8_replace_invalid_chars() to
  utf8.c that copies a string replacing any invalid UTF-8
  sequences with the Unicode specified U+FFFD replacement
  character.  For example:  "abc\xffdef" becomes "abc\uFFFDdef".
  Any UTF-8 compliant implementation will show that character
  as a � character.

* Updated res_pjsip:set_id_from_hdr() to use
  ast_utf8_replace_invalid_chars and print a warning if any
  invalid sequences were found during the copy.

* Updated stasis_channels:ast_channel_publish_varset to use
  ast_utf8_replace_invalid_chars and print a warning if any
  invalid sequences were found during the copy.

ASTERISK-27830

Change-Id: I4ffbdb19c80bf0efc675d40078a3ca4f85c567d8
2023-03-01 09:49:39 -06:00
Sean Bright e5c5cd6e25 test.c: Avoid passing -1 to FD_* family of functions.
This avoids buffer overflow errors when running tests that capture
output from child processes.

This also corrects a copypasta in an off-nominal error message.

Change-Id: Ib482847a3515364f14c7e7a0c0a4213851ddb10d
2023-02-28 10:48:11 -06:00
Naveen Albert ede67a99be chan_iax2: Fix jitterbuffer regression prior to receiving audio.
ASTERISK_29392 (a security fix) introduced a regression by
not processing frames when we don't have an audio format.

Currently, chan_iax2 only calls jb_get to read frames from
the jitterbuffer when the voiceformat has been set on the pvt.
However, this only happens when we receive a voice frame, which
means that prior to receiving voice frames, other types of frames
get stalled completely in the jitterbuffer.

To fix this, we now fallback to using the format negotiated during
call setup until we've actually received a voice frame with a format.
This ensures we're always able to read from the jitterbuffer.

ASTERISK-30354 #close
ASTERISK-30162 #close

Change-Id: Ie4fd1e8e088a145ad89e0427c2100a530e964fe9
2023-02-28 07:55:13 -06:00
Sean Bright 827222d607 test_crypto.c: Fix getcwd(…) build error.
`getcwd(…)` is decorated with the `warn_unused_result` attribute and
therefore needs its return value checked.

Change-Id: Idcccb20a0abf293202c28633d0e9ee0f6a9dbe93
2023-02-27 15:39:49 -06:00
Nick French 200dc7d0e8 pjproject_bundled: Fix cross-compilation with SSL libs.
Asterisk makefiles auto-detect SSL library availability,
then they assume that pjproject makefiles will also autodetect
an SSL library at the same time, so they do not pass on the
autodetection result to pjproject.

This normally works, except the pjproject makefiles disables
autodetection when cross-compiling.

Fix by explicitly configuring pjproject to use SSL if we
have been told to use it or it was autodetected

ASTERISK-30424 #close

Change-Id: I8fe2999ea46710e21d1d55a1bed92769c6ebded9
2023-02-27 14:48:41 -06:00
Mike Bradeen 5c11d7adea app_read: Add an option to return terminator on empty digits.
Adds 'e' option to allow Read() to return the terminator as the
dialed digits in the case where only the terminator is entered.

ie; if "#" is entered, return "#" if the 'e' option is set and ""
if it is not.

ASTERISK-30411

Change-Id: I49f3221824330a193a20c660f99da0f1fc2cbbc5
2023-02-27 13:42:39 -06:00
cmaj 5b0e3444c3 res_phoneprov.c: Multihomed SERVER cache prevention
Phones moving between subnets on multi-homed server have their
initially connected interface IP cached in the SERVER variable,
even when it is not specified in the configuration files. This
prevents phones from obtaining the correct SERVER variable value
when they move to another subnet.

ASTERISK-30388 #close
Reported-by: cmaj

Change-Id: I1d18987a9d58e85556b4c4a6814ce7006524cc92
2023-02-27 13:41:40 -06:00
Mike Bradeen 2308afed8e app_directory: Add a 'skip call' option.
Adds 's' option to skip calling the extension and instead set the
extension as DIRECTORY_EXTEN channel variable.

ASTERISK-30405

Change-Id: Ib9d9db1ba5b7524594c640461b4aa8f752db8299
2023-02-27 12:04:39 -06:00
Mike Bradeen 98742388b6 app_senddtmf: Add option to answer target channel.
Adds a new option to SendDTMF() which will answer the specified
channel if it is not already up. If no channel is specified, the
current channel will be answered instead.

ASTERISK-30422

Change-Id: Iddcbd501fcdf9fef0f453b7a8115a90b11f1d085
2023-02-27 11:13:27 -06:00
Mike Bradeen 37e558f6ef res_pjsip: Prevent SEGV in pjsip_evsub_send_request
contributed pjproject - patch to check sub->pending_notify
in evsub.c:on_tsx_state before calling
pjsip_evsub_send_request()

res_pjsip_pubsub - change post pjsip 2.13 behavior to use
pubsub_on_refresh_timeout to avoid the ao2_cleanup call on
the sub_tree. This is is because the final NOTIFY send is no
longer the last place the sub_tree is referenced.

ASTERISK-30419

Change-Id: Ib5cc662ce578e9adcda312e16c58a10b6453e438
2023-02-23 10:10:54 -06:00
Sean Bright aeb16aa7d8 app_queue: Minor docs and logging fixes for UnpauseQueueMember.
ASTERISK-30417 #close

Change-Id: I7534e7a925bf92a7b5a5347f5f54225768c162fe
2023-02-20 06:19:32 -06:00
Sean Bright aef0c0ce0e app_queue: Reset all queue defaults before reload.
Several queue fields were not being set to their default value during
a reload.

Additionally added some sample configuration options that were missing
from queues.conf.sample.

Change-Id: I3a88c7877af91752b1b46a0c087384f7eb9c47e4
2023-02-13 07:53:29 -06:00
Mike Bradeen 58636a6ea6 res_pjsip: Upgraded bundled pjsip to 2.13
Removed multiple patches.

Code chages in res_pjsip_pubsub due to changes in evsub.

Pjsip now calls on_evsub_state() before on_rx_refresh(),
so the sub tree deletion that used to take place in
on_evsub_state() now must take place in on_rx_refresh().

Additionally, pjsip now requires that you send the NOTIFY
from within on_rx_refresh(), otherwise it will assert
when going to send the 200 OK. The idea is that it will
look for this NOTIFY and cache it until after sending the
response in order to deal with the self-imposed message
mis-order. Asterisk previously dealt with this by pushing
the NOTIFY in on_rx_refresh(), but pjsip now forces us
to use it's method.

Changes were required to configure in order to detect
which way pjsip handles this as the two are not
compatible for the reasons mentioned above.

A corresponding change in testsuite is required in order
to deal with the small interal timing changes caused by
moving the NOTIFY send.

ASTERISK-30325

Change-Id: I50b00cac89d950d3511d7b250a1c641965d9fe7f
2023-02-06 18:15:35 -07:00
Sean Bright 96d9ad51ac doxygen: Fix doxygen errors.
Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85
2023-01-31 12:59:16 -06:00
Naveen Albert 88b2c741ca app_signal: Add signaling applications
Adds the Signal and WaitForSignal
applications, which can be used for inter-channel
signaling in the dialplan.

Signal supports sending a signal to other channels
listening for a signal of the same name, with an
optional data payload. The signal is received by
all channels waiting for that named signal.

ASTERISK-29810 #close

Change-Id: Ic34439de3d60f8609357666a465c354d81f5fef3
2023-01-31 05:53:37 -06:00
Mike Bradeen 70856e865f app_directory: add ability to specify configuration file
Adds option to app_directory to specify a filename from which to
read configuration instead of voicemail.conf ie;

same => n,Directory(,,c(directory.conf))

This configuration should contain a list of extensions using the
voicemail.conf format, ie;

2020=2020,Dog Dog,,,,attach=no|saycid=no|envelope=no|delete=no

ASTERISK-30404

Change-Id: Id58ccb1344ad1e563fa10db12f172fbd104a9d13
2023-01-30 09:51:06 -06:00
Naveen Albert 8a45cd7af4 func_json: Enhance parsing capabilities of JSON_DECODE
Adds support for arrays to JSON_DECODE by allowing the
user to print out entire arrays or index a particular
key or print the number of keys in a JSON array.

Additionally, adds support for recursively iterating a
JSON tree in a single function call, making it easier
to parse JSON results with multiple levels. A maximum
depth is imposed to prevent potentially blowing
the stack.

Also fixes a bug with the unit tests causing an empty
string to be printed instead of the actual test result.

ASTERISK-29913 #close

Change-Id: I603940b216a3911b498fc6583b18934011ef5d5b
2023-01-30 08:50:48 -06:00
sungtae kim f99849f8d5 res_stasis_snoop: Fix snoop crash
Added NULL pointer check and channel lock to prevent resource release
while the chanspy is processing.

ASTERISK-29604

Change-Id: Ibdc675f98052da32333b19685b1708a3751b6d24
2023-01-30 08:26:18 -06:00
Sean Bright 56051d1ac5 pbx_ael: Global variables are not expanded.
Variable references within global variable assignments are now
expanded rather than being included literally.

ASTERISK-30406 #close

Change-Id: I136e8d6395e90a4c92d9777a46a7bc3edb08d05d
2023-01-30 07:48:11 -06:00
Naveen Albert a1da8042d1 res_pjsip_session: Add overlap_context option.
Adds the overlap_context option, which can be used
to explicitly specify a context to use for overlap
dialing extension matches, rather than forcibly
using the context configured for the endpoint.

ASTERISK-30262 #close

Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db
2023-01-26 07:38:30 -06:00
Sean Bright ef16eaee36 app_playback.c: Fix PLAYBACKSTATUS regression.
In Asterisk 11, if a channel was redirected away during Playback(),
the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
(specifically commit 7d9871b394) that
behavior was inadvertently changed and the same operation would result
in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
behavior has been restored.

Partial fix for ASTERISK~25661.

Change-Id: I53f54e56b59b61c99403a481b6cb8d88b5a559ff
2023-01-13 08:28:53 -06:00
George Joseph 2f5aece0c9 res_rtp_asterisk: Don't use double math to generate timestamps
Rounding issues with double math were causing rtp timestamp
slips in outgoing packets.  We're now back to integer math
and are getting no more slips.

ASTERISK-30391

Change-Id: I6ba992b49ffdf9ebea074581dfa784a188c661a4
2023-01-12 06:00:37 -07:00
Alexei Gradinari e86d5d7fda format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...)
Each playback of WAV files results in logging
"Skipping unknown block 'LIST'".

To prevent unnecessary flooding of this DEBUG log this patch replaces
ast_log(LOG_DEBUG, ...) by ast_debug(1, ...).

Change-Id: Iaa09cf19c5348a05385518fdb8cb181b45fe05f0
2023-01-10 13:32:49 -06:00
Igor Goncharovsky 3526441e41 res_pjsip_rfc3326: Add SIP causes support for RFC3326
Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850).

ASTERISK-30319 #close

Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645
2023-01-10 13:31:14 -06:00
George Joseph 4710f37ef6 res_rtp_asterisk: Asterisk Media Experience Score (MES)
-----------------

This commit reinstates MES with some casting fixes to the
functions in time.h that convert between doubles and timeval
structures.  The casting issues were causing incorrect
timestamps to be calculated which caused transcoding from/to
G722 to produce bad or no audio.

ASTERISK-30391

-----------------

This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score.  The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics.  For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

* Updated chan_pjsip to set quality channel variables when a
  call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
  to retrieve the MES along with the existing rtcp stats when
  using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
  checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
  and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
  requested.  Also debug output that dumps the stats when an
  rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
  MES.  In the process, also had to update the calculation of
  jitter.  Many debugging statements were also changed to be
  more informative.
* Added a unit test for internal testing.  The test should not be
  run during normal operation and is disabled by default.

Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038
2023-01-09 11:40:46 -06:00
George Joseph 62ca063fca Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)"
This reverts commit d454801c2d.

Reason for revert: Issue when transcoding to/from g722

Change-Id: I09f49e171b1661548657a9ba7a978c29d0b5be86
2023-01-09 08:24:06 -06:00
Naveen Albert d33bd6d67a loader: Allow declined modules to be unloaded.
Currently, if a module declines to load, dlopen is called
to register the module but dlclose never gets called.
Furthermore, loader.c currently doesn't allow dlclose
to ever get called on the module, since it declined to
load and the unload function bails early in this case.

This can be problematic if a module is updated, since the
new module cannot be loaded into memory since we haven't
closed all references to it. To fix this, we now allow
modules to be unloaded, even if they never "loaded" in
Asterisk itself, so that dlclose is called and the module
can be properly cleaned up, allowing the updated module
to be loaded from scratch next time.

ASTERISK-30345 #close

Change-Id: Ifc743aadfa85ebe3284e02a63e124dafa64988d5
2023-01-05 06:13:25 -06:00
Naveen Albert e06fe8e344 app_broadcast: Add Broadcast application
Adds a new application, Broadcast, which can be used for
one-to-many transmission and many-to-one reception of
channel audio in Asterisk. This is similar to ChanSpy,
except it is designed for multiple channel targets instead
of a single one. This can make certain kinds of audio
manipulation more efficient and streamlined. New kinds
of audio injection impossible with ChanSpy are also made
possible.

ASTERISK-30180 #close

Change-Id: I7ba72f765dbab9b58deeae028baca3f4f8377726
2023-01-05 06:12:10 -06:00
Naveen Albert 68e345286b func_frame_trace: Print text for text frames.
Since text frames contain a text body, make FRAME_TRACE
more useful for text frames by actually printing the text.

ASTERISK-30353 #close

Change-Id: Ia6ce3d15cecd7a673a528d34faac86854a2bab50
2023-01-05 06:11:02 -06:00
Naveen Albert 3b3fef2347 json.h: Add ast_json_object_real_get.
json.h contains macros to get a string and an integer
from a JSON object. However, the macro to do this for
JSON reals is missing. This adds that.

ASTERISK-30361 #close

Change-Id: I8d0e28d763febf27b05801cdc83b73282aa6ee7a
2023-01-04 05:52:23 -06:00
Naveen Albert 7b8f7428da manager: Fix appending variables.
The if statement here is always false after the for
loop finishes, so variables are never appended.
This removes that to properly append to the end
of the variable list.

ASTERISK-30351 #close
Reported by: Sebastian Gutierrez

Change-Id: I1b7f8b85a8918f6a814cb933a479d4278cf16199
2023-01-04 05:51:18 -06:00
George Joseph 24102ba236 res_pjsip_transport_websocket: Add remote port to transport
When Asterisk receives a new websocket conenction, it creates a new
pjsip transport for it and copies connection data into it.  The
transport manager then uses the remote IP address and port on the
transport to create a monitor for each connection.  However, the
remote port wasn't being copied, only the IP address which meant
that the transport manager was creating only 1 monitoring entry for
all websocket connections from the same IP address. Therefore, if
one of those connections failed, it deleted the transport taking
all the the connections from that same IP address with it.

* We now copy the remote port into the created transport and the
  transport manager behaves correctly.

ASTERISK-30369

Change-Id: Ib506d40897ea6286455ac0be4dfbb0ed43b727e1
2023-01-03 10:53:20 -06:00
Boris P. Korzun edc90c96ac http.c: Fix NULL pointer dereference bug
If native HTTP is disabled but HTTPS is enabled and status page enabled
too, Core/HTTP crashes while loading. 'global_http_server' references
to NULL, but the status page tries to dereference it.

The patch adds a check for HTTP is enabled.

ASTERISK-30379 #close

Change-Id: I11b02fc920b72aaed9c809fc43210523ccfdc249
2023-01-03 09:55:42 -06:00
Holger Hans Peter Freyther 3d9b9a2b16 res_http_media_cache: Do not crash when there is no extension
Do not crash when a URL has no path component as in this case the
ast_uri_path function will return NULL. Make the code cope with not
having a path.

The below would crash
> media cache create http://google.com /tmp/foo.wav

Thread 1 "asterisk" received signal SIGSEGV, Segmentation fault.
0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6
(gdb) bt
 #0  0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6
 #1  0x0000ffff43d43a78 in file_extension_from_string (str=<optimized out>, buffer=buffer@entry=0xffffca9973c0 "",
    capacity=capacity@entry=64) at res_http_media_cache.c:288
 #2  0x0000ffff43d43bac in file_extension_from_url_path (bucket_file=bucket_file@entry=0x3bf96568,
    buffer=buffer@entry=0xffffca9973c0 "", capacity=capacity@entry=64) at res_http_media_cache.c:378
 #3  0x0000ffff43d43c74 in bucket_file_set_extension (bucket_file=bucket_file@entry=0x3bf96568) at res_http_media_cache.c:392
 #4  0x0000ffff43d43d10 in bucket_file_run_curl (bucket_file=0x3bf96568) at res_http_media_cache.c:555
 #5  0x0000ffff43d43f74 in bucket_http_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>)
    at res_http_media_cache.c:613
 #6  0x0000000000487638 in bucket_file_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>)
    at bucket.c:191
 #7  0x0000000000554408 in sorcery_wizard_create (object_wizard=object_wizard@entry=0x3b9f0718,
    details=details@entry=0xffffca9974a8) at sorcery.c:2027
 #8  0x0000000000559698 in ast_sorcery_create (sorcery=<optimized out>, object=object@entry=0x3bf96568) at sorcery.c:2077
 #9  0x00000000004893a4 in ast_bucket_file_create (file=file@entry=0x3bf96568) at bucket.c:727
 #10 0x00000000004f877c in ast_media_cache_create_or_update (uri=0x3bfa1103 "https://google.com",
    file_path=0x3bfa1116 "/tmp/foo.wav", metadata=metadata@entry=0x0) at media_cache.c:335
 #11 0x00000000004f88ec in media_cache_handle_create_item (e=<optimized out>, cmd=<optimized out>, a=0xffffca9976b8)
    at media_cache.c:640

ASTERISK-30375 #close

Change-Id: I6a9433688cb5d3d4be8758b7642d923bdde6c273
2023-01-03 09:37:02 -06:00
George Joseph d454801c2d res_rtp_asterisk: Asterisk Media Experience Score (MES)
This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score.  The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics.  For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

* Updated chan_pjsip to set quality channel variables when a
  call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
  to retrieve the MES along with the existing rtcp stats when
  using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
  checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
  and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
  requested.  Also debug output that dumps the stats when an
  rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
  MES.  In the process, also had to update the calculation of
  jitter.  Many debugging statements were also changed to be
  more informative.
* Added a unit test for internal testing.  The test should not be
  run during normal operation and is disabled by default.

ASTERISK-30280

Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a
2023-01-03 07:54:54 -06:00
Naveen Albert cc8d9b947b pbx_app: Update outdated pbx_exec channel snapshots.
pbx_exec makes a channel snapshot before executing applications.
This doesn't cause an issue during normal dialplan execution
where pbx_exec is called over and over again in succession.
However, if pbx_exec is called "one off", e.g. using
ast_pbx_exec_application, then a channel snapshot never ends
up getting made after the executed application returns, and
inaccurate snapshot information will linger for a while, causing
"core show channels", etc. to show erroneous info.

This is fixed by manually making a channel snapshot at the end
of ast_pbx_exec_application, since we anticipate that pbx_exec
might not get called again immediately.

ASTERISK-30367 #close

Change-Id: I2a5131053aa9d11badbc0ef2ef40b1f83d0af086
2022-12-22 09:19:55 -05:00
Naveen Albert c7598ee947 res_pjsip_session: Use Caller ID for extension matching.
Currently, there is no Caller ID available to us when
checking for an extension match when handling INVITEs.
As a result, extension patterns that depend on the Caller ID
are not matched and calls may be incorrectly rejected.

The Caller ID is not available because the supplement that
adds Caller ID to the session does not execute until after
this check. Supplement callbacks cannot yet be executed
at this point since the session is not yet in the appropriate
state.

To fix this without impacting existing behavior, the Caller ID
number is now retrieved before attempting to pattern match.
This ensures pattern matching works correctly and there is
no behavior change to the way supplements are called.

ASTERISK-28767 #close

Change-Id: Iec7f5a3b90e51b65ccf74342f96bf80314b7cfc7
2022-12-20 14:13:15 -06:00
Ben Ford 881faf544f res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.
When a call is put on hold and it has moh_passthrough and rtp_timeout
set on the endpoint, the wrong timeout will be used. rtp_timeout_hold is
expected to be used, but rtp_timeout is used instead. This change adds a
couple of checks for locally_held to determine if rtp_timeout_hold needs
to be used instead of rtp_timeout.

ASTERISK-30350

Change-Id: I7b106fc244332014216d12bba851cefe884cc25f
2022-12-20 09:38:05 -06:00
Naveen Albert 20d4775d0a app_voicemail_odbc: Fix string overflow warning.
Fixes a negative offset warning by initializing
the buffer to empty.

Additionally, although it doesn't currently complain
about it, the size of a buffer is increased to
accomodate the maximum size contents it could have.

ASTERISK-30240 #close

Change-Id: I8eecedf14d3f2a75864797f802277cac89a32877
2022-12-20 08:57:32 -06:00
Naveen Albert cbb1fd2cb9 func_callerid: Warn about invalid redirecting reason.
Currently, if a user attempts to set a Caller ID related
function to an invalid value, a warning is emitted,
except for when setting the redirecting reason.
We now emit a warning if we were unable to successfully
parse the user-provided reason.

ASTERISK-30332 #close

Change-Id: Ic341f5d5f7303b6f1115549be64db58a85944f5a
2022-12-20 08:53:25 -06:00
Igor Goncharovsky 115a1b4f0a res_pjsip: Fix path usage in case dialing with '@'
Fix aor lookup on sip path addition. Issue happens in case of dialing
with @ and overriding user part of RURI.

ASTERISK-30100 #close
Reported-by: Yury Kirsanov

Change-Id: I3f2c42a583578c94397b113e32ca3ebf2d600e13
2022-12-20 08:52:49 -06:00
Peter Fern 58404b5c22 streams: Ensure that stream is closed in ast_stream_and_wait on error
When ast_stream_and_wait returns an error (for example, when attempting
to stream to a channel after hangup) the stream is not closed, and
callers typically do not check the return code. This results in leaking
file descriptors, leading to resource exhaustion.

This change ensures that the stream is closed in case of error.

ASTERISK-30198 #close
Reported-by: Julien Alie

Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803
2022-12-20 08:51:42 -06:00
Naveen Albert 36bea9ad33 app_sendtext: Remove references to removed applications.
Removes see-also references to applications that don't
exist anymore (removed in Asterisk 19),
so these dead links don't show up on the wiki.

ASTERISK-30347 #close

Change-Id: I9539bc30f57cd65aa4e2d5ce8185eafa09567909
2022-12-20 08:19:37 -06:00
Asterisk Development Team fefc236e7c Update CHANGES and UPGRADE.txt for 20.1.0 2022-12-15 07:40:01 -05:00
Alexandre Fournier 01b3962201 res_geoloc: fix NULL pointer dereference bug
The `ast_geoloc_datastore_add_eprofile` function does not return 0 on
success, it returns the size of the underlying datastore. This means
that the datastore will be freed and its pointer set to NULL when no
error occured at all.

ASTERISK-30346

Change-Id: Iea9b209bd1244cc57b903b9496cb680c356e4bb9
2022-12-13 10:55:50 -06:00
Joshua C. Colp b6855755ce res_pjsip_aoc: Don't assume a body exists on responses.
When adding AOC to an outgoing response the code
assumed that a body would exist for comparing the
Content-Type. This isn't always true.

The code now checks to make sure the response has
a body before checking the Content-Type.

ASTERISK-21502

Change-Id: Iaead371434fc3bc693dad487228106a7d7a5ac76
2022-12-13 10:52:58 -06:00
Naveen Albert 2f9cdfbc50 app_if: Fix format truncation errors.
Fixes format truncation warnings in gcc 12.2.1.

ASTERISK-30349 #close

Change-Id: I42be4edf0284358b906e765d1966b6b9d66e1d3c
2022-12-13 08:18:08 -05:00
Michael Kuron 5c114dcb4a manager: AOC-S support for AOCMessage
ASTERISK-21502

Change-Id: I051b778f8c862d3b4794d28f2f3d782316707b08
2022-12-09 07:59:21 -06:00
Michael Kuron fee9012fe1 res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip
chan_sip supported sending AOC-D and AOC-E information in SIP INFO
messages in an "AOC" header in a format that was originally defined by
Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC
format that is supported by devices from multiple vendors, including
Snom phones with firmware >= 8.4.2 (released in 2010).

This commit adds a new res_pjsip_aoc module that inserts AOC information
into outgoing messages or sends SIP INFO messages as described below.
It also fixes a small issue in res_pjsip_session which didn't always
call session supplements on outgoing_response.

* AOC-S in the 180/183/200 responses to an INVITE request
* AOC-S in SIP INFO (if a 200 response has already been sent or if the
  INVITE was sent by Asterisk)
* AOC-D in SIP INFO
* AOC-D in the 200 response to a BYE request (if the client hangs up)
* AOC-D in a BYE request (if Asterisk hangs up)
* AOC-E in the 200 response to a BYE request (if the client hangs up)
* AOC-E in a BYE request (if Asterisk hangs up)

The specification defines one more, AOC-S in an INVITE request, which
is not implemented here because it is not currently possible in
Asterisk to have AOC data ready at this point in call setup. Once
specifying AOC-S via the dialplan or passing it through from another
SIP channel's INVITE is possible, that might be added.

The SIP INFO requests are sent out immediately when the AOC indication
is received. The others are inserted into an appropriate outgoing
message whenever that is ready to be sent. In the latter case, the XML
is stored in a channel variable at the time the AOC indication is
received. Depending on where the AOC indications are coming from (e.g.
PRI or AMI), it may not always be possible to guarantee that the AOC-E
is available in time for the BYE.

Successfully tested AOC-D and both variants of AOC-E with a Snom D735
running firmware 10.1.127.10. It does not appear to properly support
AOC-S however, so that could only be tested by inspecting SIP traces.

ASTERISK-21502 #close
Reported-by: Matt Jordan <mjordan@digium.com>

Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333
2022-12-09 07:57:21 -06:00
Joshua C. Colp 564349ff5d ari: Destroy body variables in channel create.
When passing a JSON body to the 'create' channel route
it would be converted into Asterisk variables, but never
freed resulting in a memory leak.

This change makes it so that the variables are freed in
all cases.

ASTERISK-30344

Change-Id: I924dbd866a01c6073e2d6fb846ccaa27ef72d49d
2022-12-09 06:48:54 -06:00
Naveen Albert b9c031c1f8 app_voicemail: Fix missing email in msg_create_from_file.
msg_create_from_file currently does not dispatch emails,
which means that applications using this function, such
as MixMonitor, will not trigger notifications to users
(only AMI events are sent our currently). This is inconsistent
with other ways users can receive voicemail.

This is fixed by adding an option that attempts to send
an email and falling back to just the notifications as
done now if that fails. The existing behavior remains
the default.

ASTERISK-30283 #close

Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7
2022-12-09 06:45:16 -06:00
Marcel Wagner 58534b309f res_pjsip: Fix typo in from_domain documentation
This fixes a small typo in the from_domain documentation on the endpoint documentation

ASTERISK-30328 #close

Change-Id: Ia6f0897c3f5cab899ef2cde6b3ac07265b8beb21
2022-12-09 06:44:07 -06:00
Naveen Albert 531eacd6c9 res_hep: Add support for named capture agents.
Adds support for the capture agent name field
of the Homer protocol to Asterisk by allowing
users to specify a name that will be sent to
the HEP server.

ASTERISK-30322 #close

Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b
2022-12-08 21:31:42 -06:00
Naveen Albert b365ea8601 app_if: Adds conditional branch applications
Adds the If, ElseIf, Else, ExitIf, and EndIf
applications for conditional execution
of a block of dialplan, similar to the While,
EndWhile, and ExitWhile applications. The
appropriate branch is executed at most once
if available and may be broken out of while
inside.

ASTERISK-29497

Change-Id: I3aa3bd35a5add82465c6ee9bd86b64601f0e1f49
2022-12-08 13:57:57 -06:00
Naveen Albert 0d6003fa9a res_pjsip_session.c: Map empty extensions in INVITEs to s.
Some SIP devices use an empty extension for PLAR functionality.

Rather than rejecting these empty extensions, we now use the s
extension for such calls to mirror the existing PLAR functionality
in Asterisk (e.g. chan_dahdi).

ASTERISK-30265 #close

Change-Id: I0861a405cd49bbbf532b52f7b47f0e2810832590
2022-12-08 13:56:38 -06:00
Marcel Wagner b83af13f65 res_pjsip: Update contact_user to point out default
Updates the documentation for the 'contact_user' field to point out the
default outbound contact if no contact_user is specified 's'

ASTERISK-30316 #close

Change-Id: I61f24fb9164e4d07e05908a2511805281874c876
2022-12-08 12:39:46 -06:00
Naveen Albert 80e6205bb0 res_adsi: Fix major regression caused by media format rearchitecture.
The commit that rearchitected media formats,
a2c912e997 (ASTERISK_23114)
introduced a regression by improperly translating code in res_adsi.c.
In particular, the pointer to the frame buffer was initialized
at the top of adsi_careful_send, rather than dynamically updating it
for each frame, as is required.

This resulted in the first frame being repeatedly sent,
rather than advancing through the frames.
This corrupted the transmission of the CAS to the CPE,
which meant that CPE would never respond with the DTMF acknowledgment,
effectively completely breaking ADSI functionality.

This issue is now fixed, and ADSI now works properly again.

ASTERISK-29793 #close

Change-Id: Icdeddf733eda2981c98712d1ac9cddc0db507dbe
2022-12-08 12:37:12 -06:00
Naveen Albert 406143ae61 res_pjsip_header_funcs: Add custom parameter support.
Adds support for custom URI and header parameters
in the From header in PJSIP. Parameters can be
both set and read using this function.

ASTERISK-30150 #close

Change-Id: Ifb1bc3c512ad5f6faeaebd7817f004a2ecbd6428
2022-12-08 12:25:26 -06:00
Naveen Albert 83eb113e0f func_presencestate: Fix invalid memory access.
When parsing information from AstDB while loading,
it is possible that certain pointers are never
set, which leads to invalid memory access and
then, fatally, invalid free attempts on this memory.
We now initialize to NULL to prevent this.

ASTERISK-30311 #close

Change-Id: I6120681d04fd2c12a9473f35ce95a1f8e74e3929
2022-12-08 10:21:26 -06:00
Naveen Albert b90e57758b sig_analog: Fix no timeout duration.
ASTERISK_28702 previously attempted to fix an
issue with flash hook hold timing out after
just under 17 minutes, when it should have never
been timing out. It fixed this by changing 999999
to INT_MAX, but it did so in chan_dahdi, which
is the wrong place since ss_thread is now in
sig_analog and the one in chan_dahdi is mostly
dead code.

This fixes this by porting the fix to sig_analog.

ASTERISK-30336 #close

Change-Id: I05eb69cc0b5319d357842a70bd26ef64d145cb15
2022-12-08 10:13:47 -06:00
Naveen Albert 52c7d3ed07 xmldoc: Allow XML docs to be reloaded.
The XML docs are currently only loaded on
startup with no way to update them during runtime.
This makes it impossible to load modules that
use ACO/Sorcery (which require documentation)
if they are added to the source tree and built while
Asterisk is running (e.g. external modules).

This adds a CLI command to reload the XML docs
during runtime so that documentation can be updated
without a full restart of Asterisk.

ASTERISK-30289 #close

Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07
2022-12-08 09:17:26 -06:00
Naveen Albert a4bcdce1db rtp_engine.h: Update examples using ast_format_set.
This file includes some doxygen comments referencing
ast_format_set. This is an obsolete API that was
removed years back, but documentation was not fully
updated to reflect that. These examples are
updated to the current way of doing things
(using the format cache).

ASTERISK-30327 #close

Change-Id: I570f3b8007fa17ba470cc7117f44bfe7c555d2f7
2022-12-08 09:17:18 -06:00
Naveen Albert 691178c48e app_mixmonitor: Add option to use real Caller ID for voicemail.
MixMonitor currently uses the Connected Line as the Caller ID
for voicemails. This is due to the implementation being written
this way for use with Digium phones. However, in general this
is not correct for generic usage in the dialplan, and people
may need the real Caller ID instead. This adds an option to do that.

ASTERISK-30286 #close

Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c
2022-12-08 08:04:35 -06:00
Ben Ford d476994768 pjproject: 2.13 security fixes
Backports two security fixes (c4d3498 and 450baca) from pjproject 2.13.

ASTERISK-30338

Change-Id: I86fdc003d5d22cb66e7cc6dc3313a8194f27eb69
2022-12-03 10:26:40 -06:00
George Joseph 7684c9e907 pjsip_transport_events: Fix possible use after free on transport
It was possible for a module that registered for transport monitor
events to pass in a pjsip_transport that had already been freed.
This caused pjsip_transport_events to crash when looking up the
monitor for the transport.  The fix is a two pronged approach.

1. We now increment the reference count on pjsip_transports when we
create monitors for them, then decrement the count when the
transport is going to be destroyed.

2. There are now APIs to register and unregister monitor callbacks
by "transport key" which is a string concatenation of the remote ip
address and port.  This way the module needing to monitor the
transport doesn't have to hold on to the transport object itself to
unregister.  It just has to save the transport_key.

* Added the pjsip_transport reference increment and decrement.

* Changed the internal transport monitor container key from the
  transport->obj_name (which may not be unique anyway) to the
  transport_key.

* Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that
  fills a buffer with the transport_key using a passed-in
  pjsip_transport.

* Added the following functions:
  ast_sip_transport_monitor_register_key
  ast_sip_transport_monitor_register_replace_key
  ast_sip_transport_monitor_unregister_key
  and marked their non-key counterparts as deprecated.

* Updated res_pjsip_pubsub and res_pjsip_outbound_register to use
  the new "key" monitor functions.

NOTE: res_pjsip_registrar also uses the transport monitor
functionality but doesn't have a persistent object other than
contact to store a transport key.  At this time, it continues to
use the non-key monitor functions.

ASTERISK-30244

Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b
2022-12-03 10:24:36 -06:00
Mike Bradeen 81f10e847e manager: prevent file access outside of config dir
Add live_dangerously flag to manager and use this flag to
determine if a configuation file outside of AST_CONFIG_DIR
should be read.

ASTERISK-30176

Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404
2022-12-03 10:22:18 -06:00
Mike Bradeen eb1d7ab53c ooh323c: not checking for IE minimum length
When decoding q.931 encoded calling/called number
now checking for length being less than minimum required.

ASTERISK-30103

Change-Id: I3dcfce0f35eca258dc450f87c92d4d7af402c2e7
2022-12-01 11:11:42 -06:00
Naveen Albert c7df5ee7c1 pbx_builtins: Allow Answer to return immediately.
The Answer application currently waits for up to 500ms
for media, even if users specify a different timeout.

This adds an option to not wait for media on the channel
by doing a raw answer instead. The default 500ms threshold
is also documented.

ASTERISK-30308 #close

Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4
2022-11-29 09:23:49 -06:00
Naveen Albert 5ede4e217a chan_dahdi: Allow FXO channels to start immediately.
Currently, chan_dahdi will wait for at least one
ring before an incoming call can enter the dialplan.
This is generally necessary in order to receive
the Caller ID spill and/or distinctive ringing
detection.

However, if neither of these is required, then there
is nothing gained by waiting for one ring and this
unnecessarily delays call setup. Users can now
use immediate=yes to make FXO channels (FXS signaled)
begin processing dialplan as soon as Asterisk receives
the call.

ASTERISK-30305 #close

Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5
2022-11-29 08:29:21 -06:00
Maximilian Fridrich 60b81eabe0 core & res_pjsip: Improve topology change handling.
This PR contains two relatively separate changes in channel.c and
res_pjsip_session.c which ensure that topology changes are not ignored
in cases where they should be handled.

For channel.c:

The function ast_channel_request_stream_topology_change only triggers a
stream topology request change indication, if the channel's topology
does not equal the requested topology. However, a channel could be in a
state where it is currently "negotiating" a new topology but hasn't
updated it yet, so the topology request change would be lost. Channels
need to be able to handle such situations internally and stream
topology requests should therefore always be passed on.

In the case of chan_pjsip for example, it queues a session refresh
(re-INVITE) if it is currently in the middle of a transaction or has
pending requests (among other reasons).

Now, ast_channel_request_stream_topology_change always indicates a
stream topology request change even if the requested topology equals the
channel's topology.

For res_pjsip_session.c:

The function resolve_refresh_media_states does not process stream state
changes if the delayed active state differs from the current active
state. I.e. if the currently active stream state has changed between the
time the sip session refresh request was queued and the time it is being
processed, the session refresh is ignored. However, res_pjsip_session
contains logic that ensures that session refreshes are queued and
re-queued correctly if a session refresh is currently not possible. So
this check is not necessary and led to some session refreshes being
lost.

Now, a session refresh is done even if the delayed active state differs
from the current active state and it is checked whether the delayed
pending state differs from the current active - because that means a
refresh is necessary.

Further, the unit test of resolve_refresh_media_states was adapted to
reflect the new behavior. I.e. the changes to delayed pending are
prioritized over the changes to current active because we want to
preserve the original intention of the pending state.

ASTERISK-30184

Change-Id: Icd0703295271089057717006730b555b9a1d4e5a
2022-11-29 08:23:49 -06:00
Naveen Albert 2efa290d3c sla: Prevent deadlock and crash due to autoservicing.
SLAStation currently autoservices the station channel before
creating a thread to actually dial the trunk. This leads
to duplicate servicing of the channel which causes assertions,
deadlocks, crashes, and moreover not the correct behavior.

Removing the autoservice prevents the crash, but if the station
hangs up before the trunk answers, the call hangs since the hangup
was never serviced on the channel.

This is fixed by not autoservicing the channel, but instead
servicing it in the thread dialing the trunk, since it is doing
so synchronously to begin with. Instead of sleeping for 100ms
in a loop, we simply use the channel for timing, and abort
if it disappears.

The same issue also occurs with SLATrunk when a call is answered,
because ast_answer invokes ast_waitfor_nandfds. Thus, we use
ast_raw_answer instead which does not cause any conflict and allows
the call to be answered normally without thread blocking issues.

ASTERISK-29998 #close

Change-Id: Icc237d50354b5910000d2305901e86d2c87bb9d8
2022-11-28 08:54:30 -06:00
Jaco Kroon ce2153fc5a Build system: Avoid executable stack.
Found in res_geolocation, but I believe others may have similar issues,
thus not linking to a specific issue.

Essentially gcc doesn't mark the stack for being non-executable unless
it's compiling the source, this informs ld via gcc to mark the object as
not requiring an executable stack (which a binary blob obviously
doesn't).

ASTERISK-30321

Change-Id: I71bcc2fd1fe0c82a28b3257405d6f2b566fd9bfc
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
2022-11-21 08:52:49 -06:00
Naveen Albert 002afc3f2a func_json: Fix memory leak.
A memory leak was present in func_json due to
using ast_json_free, which just calls ast_free,
as opposed to recursively freeing the JSON
object as needed. This is now fixed to use the
right free functions.

ASTERISK-30293 #close

Change-Id: I982324dde841dc9147c8d8ad35c8719daf418b49
2022-11-21 08:29:40 -06:00
Naveen Albert 1e77b8c473 test_json: Remove duplicated static function.
Removes the function mkstemp_file and uses
ast_file_mkftemp from file.h instead.

ASTERISK-30295 #close

Change-Id: I7412ec06f88c39ee353bcdb8c976c2fcac546609
2022-11-21 07:43:38 -06:00
Joshua C. Colp 61922d2934 res_agi: Respect "transmit_silence" option for "RECORD FILE".
The "RECORD FILE" command in res_agi has its own
implementation for actually doing the recording. This
has resulted in it not actually obeying the option
"transmit_silence" when recording.

This change causes it to now send silence if the
option is enabled.

ASTERISK-30314

Change-Id: Ib3a85601ff35d1b904f836691bad8a4b7e957174
2022-11-16 06:43:41 -05:00
Naveen Albert 6e59b01e1a app_mixmonitor: Add option to delete files on exit.
Adds an option that allows MixMonitor to delete
its copy of any recording files before exiting.

This can be handy in conjunction with options
like m, which copy the file elsewhere, and the
original files may no longer be needed.

ASTERISK-30284 #close

Change-Id: Ida093679c67e300efc154a97b6d8ec0f104e581e
2022-11-08 13:46:50 -06:00
Naveen Albert 49cfdbbdff manager: Update ModuleCheck documentation.
The ModuleCheck XML documentation falsely
claims that the module's version number is returned.
This has not been the case since 14, since the version
number is not available anymore, but the documentation
was not changed at the time. It is now updated to
reflect this.

ASTERISK-30285 #close

Change-Id: Idde2d1205a11f2623fa1ddab192faa3dc4081e91
2022-11-08 08:16:53 -06:00
Naveen Albert 8142b313c3 file.c: Don't emit warnings on winks.
Adds an ignore case for wink since it should
pass through with no warning.

ASTERISK-30290 #close

Change-Id: Ieb7e34daa717357ac5c93efb0059f6c2321f16ad
2022-11-06 11:51:02 -05:00
George Joseph 0c1c623dee runUnittests.sh: Save coredumps to proper directory
Fixed the specification of "outputdir" when calling ast_coredumper
so the txt files are saved in the correct place.

ASTERISK-30282

Change-Id: Ic631cb90c1e4c29d970c982dff45fda5e0eb15b6
2022-11-02 12:02:55 -05:00
Naveen Albert dfe2f38642 app_stack: Print proper exit location for PBXless channels.
When gosub is executed on channels without a PBX, the context,
extension, and priority are initialized to the channel driver's
default location for that endpoint. As a result, the last Return
will restore this location and the Gosub logs will print out bogus
information about our exit point.

To fix this, on channels that don't have a PBX, the execution
location is left intact on the last return if there are no
further stack frames left. This allows the correct location
to be printed out to the user, rather than the bogus default
context.

ASTERISK-30076 #close

Change-Id: I1d42a99c9aa9e3708d32718863175158a894e414
2022-11-02 10:50:27 -05:00
George Joseph f723b465e5 chan_rtp: Make usage of ast_rtp_instance_get_local_address clearer
unicast_rtp_request() was setting the channel variables like this:

pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS",
    ast_sockaddr_stringify_addr(&local_address));
ast_rtp_instance_get_local_address(instance, &local_address);
pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT",
    ast_sockaddr_stringify_port(&local_address));

...which made it appear that UNICASTRTP_LOCAL_ADDRESS was being
set before local_address was set.  In fact, the address part of
local_address was set earlier in the function, just not the port.
This was confusing however so ast_rtp_instance_get_local_address()
is now being called before setting UNICASTRTP_LOCAL_ADDRESS.

ASTERISK-30281

Change-Id: I872ac49477100f4eb33891d46efc6ca21ec81aa4
2022-11-02 08:55:51 -05:00
Mike Bradeen 50e2921a48 res_pjsip: prevent crash on websocket disconnect
When a websocket (or potentially any stateful connection) is quickly
created then destroyed, it is possible that the qualify thread will
destroy the transaction before the initialzing thread is finished
with it.

Depending on the timing, this can cause an assertion within pjsip.

To prevent this, ast_send_stateful_response will now create the group
lock and add a reference to it before creating the transaction.

While this should resolve the crash, there is still the potential that
the contact will not be cleaned up properly, see:ASTERISK~29286. As a
result, the contact has to 'time out' before it will be removed.

ASTERISK-28689

Change-Id: Id050fded2247a04d8f0fc5b8a2cf3e5482cb8cee
2022-10-31 10:09:39 -05:00
Naveen Albert afd86b47c1 tcptls: Prevent crash when freeing OpenSSL errors.
write_openssl_error_to_log has been erroneously
using ast_free instead of free, which will
cause a crash when MALLOC_DEBUG is enabled since
the memory was not allocated by Asterisk's memory
manager. This changes it to use the actual free
function directly to avoid this.

ASTERISK-30278 #close

Change-Id: Iac8b6468b718075809c45d8ad16b101af21a474d
2022-10-31 09:41:52 -05:00
Igor Goncharovsky 096529d33f res_pjsip_outbound_registration: Allow to use multiple proxies for registration
Current registration code use pjsip_parse_uri to verify outbound_proxy
that is different from the reading this option for the endpoint. This
made value with multiple proxies invalid for registration pjsip settings.
Removing URI validation helps to use registration through multiple proxies.

ASTERISK-30217 #close

Change-Id: I064558e66f04b9f3260c46181812a01349761357
2022-10-28 11:38:41 -05:00
Naveen Albert ca8900b0f6 tests: Fix compilation errors on 32-bit.
Fix compilation errors caused by using size_t
instead of uintmax_t and non-portable format
specifiers.

ASTERISK-30273 #close

Change-Id: I363e6057ef84d54b88af80d23ad6147eef9216ee
2022-10-27 14:29:45 -05:00
Henning Westerholt 12445040d3 res_pjsip: return all codecs on a re-INVITE without SDP
Currently chan_pjsip on receiving a re-INVITE without SDP will only
return the codecs that are previously negotiated and not offering
all enabled codecs.

This causes interoperability issues with different equipment (e.g.
from Cisco) for some of our customers and probably also in other
scenarios involving 3PCC infrastructure.

According to RFC 3261, section 14.2 we SHOULD return all codecs
on a re-INVITE without SDP

The PR proposes a new parameter to configure this behaviour:
all_codecs_on_empty_reinvite. It includes the code, documentation,
alembic migrations, CHANGES file and example configuration additions.

ASTERISK-30193 #close

Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148
2022-10-27 11:22:20 -05:00
Naveen Albert 40b52322e5 res_pjsip_notify: Add option support for AMI.
The PJSIP notify CLI commands allow for using
"options" configured in pjsip_notify.conf.

This allows these same options to be used in
AMI actions as well.

Additionally, as part of this improvement,
some repetitive common code is refactored.

ASTERISK-30263 #close

Change-Id: Ie4496b322b63b61eaf9672183a959ab99a04b6b5
2022-10-27 10:07:20 -05:00
Naveen Albert c32b39d123 res_pjsip_logger: Add method-based logging option.
Expands the pjsip logger to support the ability to filter
by SIP message method. This can make certain types of SIP debugging
easier by only logging messages of particular method(s).

ASTERISK-30146 #close

Co-authored-by: Sean Bright <sean@seanbright.com>
Change-Id: I9c8cbb6fc8686ef21190eb42e08bc9a9b147707f
2022-10-27 09:00:29 -05:00
Frederic LE FOLL 50a4495799 Dialing API: Cancel a running async thread, may not cancel all calls
race condition: ast_dial_join() may not cancel outgoing call, if
function is called just after called party answer and before
application execution (bit is_running_app not yet set).

This fix adds ast_softhangup() calls in addition to existing
pthread_kill() when is_running_app is not set.

ASTERISK-30258

Change-Id: Idbdd5c15122159661aa8e996a42d5800083131e4
2022-10-27 07:52:12 -05:00
Naveen Albert 180ca32565 chan_dahdi: Fix unavailable channels returning busy.
This fixes dahdi_request to properly set the cause
code to CONGESTION instead of BUSY if no channels
were actually available.

Currently, the cause is erroneously set to busy
if the channel itself is found, regardless of its
current state. However, if the channel is not available
(e.g. T1 down, card not operable, etc.), then the
channel itself may not be in a functional state,
in which case CHANUNAVAIL is the correct cause to use.

This adds a simple check to ensure that busy tone
is only returned if a channel is encountered that
has an owner, since that is the only possible way
that a channel could actually be busy.

ASTERISK-30274 #close

Change-Id: Iad5870223c081240c925b19df8d6af136953b994
2022-10-26 11:14:25 -05:00
Naveen Albert 9258d8212a res_pjsip_pubsub: Prevent removing subscriptions.
pjproject does not provide any mechanism of removing
event packages, which means that once a subscription
handler is registered, it is effectively permanent.

pjproject will assert if the same event package is
ever registered again, so currently unloading and
loading any Asterisk modules that use subscriptions
will cause a crash that is beyond our control.

For that reason, we now prevent users from being
able to unload these modules, to prevent them
from ever being loaded twice.

ASTERISK-30264 #close

Change-Id: I7fdcb1a5e44d38b7ba10c44259fe98f0ae9bc12c
2022-10-26 09:08:17 -05:00
Naveen Albert 407216a0a5 say: Don't prepend ampersand erroneously.
Some logic in say.c for determining if we need
to also add an ampersand for file seperation was faulty,
as non-successful files would increment the count, causing
a leading ampersand to be added improperly.

This is fixed, and a unit test that captures this regression
is also added.

ASTERISK-30248 #close

Change-Id: I02c1d3a11d82fe4ea8b462070cbd1effb5834d2b
2022-10-26 07:48:17 -05:00
Philip Prindeville d0bea5a725 res_crypto: handle unsafe private key files
ASTERISK-30213 #close

Change-Id: I4a77143d41615b7c4fc25bb1251c0a9cb87b417a
2022-10-14 10:01:06 -05:00
Mike Bradeen 907d7e7d7d audiohook: add directional awareness
Add enum to allow setting optional direction. If set to only one
direction, only feed matching-direction frames to the associated
slin factory.

This prevents mangling the transcoder on non-mixed frames when the
READ and WRITE frames would have otherwise required it.  Also
removes the need to mute or discard the un-wanted frames as they
are no longer added in the first place.

res_stasis_snoop is changed to use this addition to set direction
on audiohook based on spy direction.

If no direction is set, the ast_audiohook_init will init this enum
to BOTH which maintains existing functionality.

ASTERISK-30252

Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb
2022-10-11 08:13:18 -05:00
Naveen Albert b331caca30 cdr: Allow bridging and dial state changes to be ignored.
Allows bridging, parking, and dial messages to be globally
ignored for all CDRs such that only a single CDR record
is generated per channel.

This is useful when CDRs should endure for the lifetime of
an entire channel and bridging and dial updates in the
dialplan should not result in multiple CDR records being
created for the call. With the ignore bridging option,
bridging changes have no impact on the channel's CDRs.
With the ignore dial state option, multiple Dials and their
outcomes have no impact on the channel's CDRs. The
last disposition on the channel is preserved in the CDR,
so the actual disposition of the call remains available.

These two options can reduce the amount of "CDR hacks" that
have hitherto been necessary to ensure that CDR was not
"spoiled" by these messages if that was undesired, such as
putting a dummy optimization-disabled local channel between
the caller and the actual call and putting the CDR on the channel
in the middle to ensure that CDR would persist for the entire
call and properly record start, answer, and end times.
Enabling these options is desirable when calls correspond
to the entire lifetime of channels and the CDR should
reflect that.

Current default behavior remains unchanged.

ASTERISK-30091 #close

Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832
2022-10-10 12:06:36 -05:00
Naveen Albert e0e7f35730 res_tonedetect: Add ringback support to TONE_DETECT.
Adds support for detecting audible ringback tone
to the TONE_DETECT function using the p option.

ASTERISK-30254 #close

Change-Id: Ie2329ff245248768367d26749c285fbe823f6414
2022-10-10 12:04:33 -05:00
Naveen Albert 98fc05f13b chan_dahdi: Resolve format truncation warning.
Fixes a format truncation warning in notify_message.

ASTERISK-30256 #close

Change-Id: I983a423c0214641ca4f8c9dfe0b19c47448fdee1
2022-10-10 10:31:13 -05:00
Philip Prindeville ef74ecacc7 res_crypto: don't modify fname in try_load_key()
"fname" is passed in as a const char *, but strstr() mangles that
into a char *, and we were attempting to modify the string in place.
This is an unwanted (and undocumented) side-effect.

ASTERISK-30213

Change-Id: Ifa36d352aafeb7f9beec3f746332865c7d21e629
2022-10-10 10:13:41 -05:00
Philip Prindeville 5e2485b5c0 res_crypto: use ast_file_read_dirs() to iterate
ASTERISK-30213

Change-Id: I115f5f8942ffcfb23cd2559a55bac8a2eba081e0
2022-10-10 10:11:15 -05:00
George Joseph 2a500b325a res_geolocation: Update wiki documentation
Also added a note to the geolocation.conf.sample file
and added a README to the res/res_geolocation/wiki
directory.

Change-Id: I89c3c5db8c0701b33127993622d5e4f904bddfbc
2022-10-10 07:31:43 -05:00
Maximilian Fridrich 0d2e140123 res_pjsip: Add mediasec capabilities.
This patch adds support for mediasec SIP headers and SDP attributes.
These are defined in RFC 3329, 3GPP TS 24.229 and
draft-dawes-sipcore-mediasec-parameter. The new features are
implemented so that a backbone for RFC 3329 is present to streamline
future work on RFC 3329.

With this patch, Asterisk can communicate with Deutsche Telekom trunks
which require these fields.

ASTERISK-30032

Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2
2022-09-29 04:10:48 -05:00
Asterisk Development Team 7f80830ced Update CHANGES and UPGRADE.txt for 20.0.0 2022-09-28 07:44:57 -05:00
Holger Hans Peter Freyther 62881c668b res_prometheus: Do not crash on invisible bridges
Avoid crashing by skipping invisible bridges and checking the
snapshot for a null pointer. In effect this is how the bridges
are enumerated in res/ari/resource_bridges.c already.

ASTERISK-30239
ASTERISK-30237

Change-Id: I58ef9f44036feded5966b5fc70ae754f8182883d
2022-09-26 19:27:58 -05:00
Naveen Albert 8afb313a43 res_pjsip_geolocation: Change some notices to debugs.
If geolocation is not in use for an endpoint, the NOTICE
log level is currently spammed with messages about this,
even though nothing is wrong and these messages provide
no real value. These log messages are therefore changed
to debugs.

ASTERISK-30241 #close

Change-Id: I656b355d812f67cc0f0fdf09b00b0e1458598bb4
2022-09-26 15:03:32 -05:00
Naveen Albert 7335b0cffe db: Fix incorrect DB tree count for AMI.
The DBGetTree AMI action's ListItem previously
always reported 1, regardless of the count. This
is corrected to report the actual count.

ASTERISK-30245 #close
patches:
  gettreecount.diff submitted by Birger Harzenetter (license 5870)

Change-Id: I46d8992710f1b8524426b1255f57d1ef4a4934d4
2022-09-26 14:11:17 -05:00
Naveen Albert 407167cc28 func_logic: Don't emit warning if both IF branches are empty.
The IF function currently emits warnings if both IF branches
are empty. However, there is no actual necessity that either
branch be non-empty as, unlike other conditional applications/
functions, nothing is inherently done with IF, and both
sides could legitimately be empty. The warning is thus turned
into a debug message.

ASTERISK-30243 #close

Change-Id: I5250625dd720f95e1859b5dfb933905d7e7a730e
2022-09-26 12:20:21 -05:00
Naveen Albert a5ec60e6c6 features: Add no answer option to Bridge.
Adds the n "no answer" option to the Bridge application
so that answer supervision can not automatically
be provided when Bridge is executed.

Additionally, a mechanism (dialplan variable)
is added to prevent bridge targets (typically the
target of a masquerade) from answering the channel
when they enter the bridge.

ASTERISK-30223 #close

Change-Id: I76f73fcd8e403bcd18f2abb40c658f537ac1ba6d
2022-09-26 11:44:20 -05:00
Naveen Albert 1e29607b5c app_bridgewait: Add option to not answer channel.
Adds the n option to not answer the channel when calling
BridgeWait, so the application can be used without
forcing answer supervision.

ASTERISK-30216 #close

Change-Id: I6b85ef300b1f7b5170f8537e2b10889cc2e6605a
2022-09-26 10:41:46 -05:00
Naveen Albert 8c791f9a65 app_amd: Add option to play audio during AMD.
Adds an option that will play an audio file
to the party while AMD is running on the
channel, so the called party does not just
hear silence.

ASTERISK-30179 #close

Change-Id: I4af306274552b61b3d9f0883c33f698abd4699b6
2022-09-26 09:43:14 -05:00
Philip Prindeville 3e7ce90f9c test: initialize capture structure before freeing
ASTERISK-30232 #close

Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902
2022-09-26 09:40:26 -05:00
Naveen Albert 1ed4518328 func_export: Add EXPORT function
Adds the EXPORT function, which allows write
access to variables and functions on other
channels.

ASTERISK-29432 #close

Change-Id: I7492645ae4307553d0f586d78e13a4f586231fdf
2022-09-26 07:53:20 -05:00
Maximilian Fridrich 5bbad0d27c res_pjsip: Add 100rel option "peer_supported".
This patch adds a new option to the 100rel parameter for pjsip
endpoints called "peer_supported". When an endpoint with this option
receives an incoming request and the request indicated support for the
100rel extension, then Asterisk will send 1xx responses reliably. If
the request did not indicate 100rel support, Asterisk sends 1xx
responses normally.

ASTERISK-30158

Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad
2022-09-22 18:39:50 -05:00
Naveen Albert 8aae0b9f08 func_scramble: Fix null pointer dereference.
Fix segfault due to null pointer dereference
inside the audiohook callback.

ASTERISK-30220 #close

Change-Id: Ideb80f606974366e89d619d908744230b5a5a259
2022-09-22 11:26:09 -05:00
Jaco Kroon 278c5726ca manager: be more aggressive about purging http sessions.
If we find that n_max (currently hard wired to 1) sessions were purged,
schedule the next purge for 1ms into the future rather than 5000ms (as
per current).  This way we will purge up to 1000 sessions per second
rather than 1 every 5 seconds.

This mitigates a build-up of sessions should http sessions gets
established faster than 1 per 5 seconds.

Change-Id: I9820d39aa080109df44fe98c1325cafae48d54f5
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
2022-09-22 11:11:40 -05:00
Naveen Albert ab1dbfef75 func_strings: Add trim functions.
Adds TRIM, LTRIM, and RTRIM, which can be used
for trimming leading and trailing whitespace
from strings.

ASTERISK-30222 #close

Change-Id: I50fb0c40726d044a7a41939fa9026f3da4872554
2022-09-22 05:49:00 -05:00
George Joseph e25b690d10 res_crypto: Memory issues and uninitialized variable errors
ASTERISK-30235

Change-Id: Ia1e326e7b52cd06fd5e6c9009e3e63193c92f6cd
2022-09-19 05:32:32 -06:00
George Joseph e33f2dcc0f res_geolocation: Fix issues exposed by compiling with -O2
Fixed "may be used uninitialized" errors in geoloc_config.c.

ASTERISK-30234

Change-Id: I1ea336bf7abbc16fa59b75720f0db8f1d960b3d4
2022-09-16 08:42:26 -06:00
Philip Prindeville 026dc08529 res_crypto: don't complain about directories
ASTERISK-30226 #close

Change-Id: I5695fb0c9521f112f754b8362cff2a8f3eff05c5
2022-09-14 23:16:37 -06:00
Asterisk Development Team f01ed3eea4 Update CHANGES and UPGRADE.txt for 20.0.0 2022-09-14 09:25:44 -05:00
Mike Bradeen 7a44296ca9 res_pjsip: Add user=phone on From and PAID for usereqphone=yes
Adding user=phone to local-side uri's when user_eq_phone=yes is set for
an endpoint. Previously this would only add the header to the To and R-URI.

ASTERISK-30178

Change-Id: Id3bfb5d225d762e7d2668c023fe09e4541ae8600
2022-09-14 07:20:22 -05:00
George Joseph 8cbea1c7ef res_geolocation: Fix segfault when there's an empty element
Fixed a segfault caused by var_list_from_loc_info() encountering
an empty location info element.

Fixed an issue in ast_strsep() where a value with only whitespace
wasn't being preserved.

Fixed an issue in ast_variable_list_from_quoted_string() where
an empty value was considered a failure.

ASTERISK-30215
Reported by: Dan Cropp

Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a
2022-09-13 09:51:25 -05:00
sungtae kim 80bc844fd6 res_musiconhold: Add option to not play music on hold on unanswered channels
This change adds an option, answeredonly, that will prevent music on
hold on channels that are not answered.

ASTERISK-30135

Change-Id: I1ab0defa43a29a26ae39f94c623596cf90fddc08
2022-09-13 05:46:48 -05:00
Ben Ford 881a3f2306 res_pjsip: Add TEL URI support for basic calls.
This change allows TEL URI requests to come through for basic calls. The
allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To
headers will now allow TEL URIs, as well as the request URI.

Support is only for TEL URIs present in traffic from a remote party.
Asterisk does not generate any TEL URIs on its own.

ASTERISK-26894

Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a
2022-09-13 04:51:00 -05:00
Philip Prindeville 3e054c9ebc res_crypto: Use EVP API's instead of legacy API's
ASTERISK-30046 #close

Change-Id: I5c738756de75fd27ebad54be144c0ac6193f21b2
2022-09-12 16:19:37 -05:00
Philip Prindeville 736cdf84f4 test: Add coverage for res_crypto
We're validating the following functionality:

encrypting a block of data with RSA
decrypting a block of data with RSA
signing a block of data with RSA
verifying a signature with RSA
encrypting a block of data with AES-ECB
encrypting a block of data with AES-ECB

as well as accessing test keys from the keystore.

ASTERISK-30045 #close

Change-Id: I0d10e7b41009c5290a4356c6480e636712d5c96d
2022-09-12 16:18:19 -05:00
Philip Prindeville 2d7656cb50 res_crypto: make keys reloadable on demand for testing
ASTERISK-30045

Change-Id: If59bbb50c1771084bfe2fef307a6077c90d35ce8
2022-09-12 13:08:49 -05:00
Philip Prindeville 5809d879b0 test: Add test coverage for capture child process output
ASTERISK-30037 #close

Change-Id: I0273e85eeeb6b8e46703f24cd74d84f3daf0a69a
2022-09-12 11:23:04 -05:00
Philip Prindeville 2c4c44ca64 main/utils: allow checking for command in $PATH
ASTERISK-30037

Change-Id: I4b6f7264c8c737c476c798d2352f3232b263bbdf
2022-09-12 09:49:21 -05:00
Philip Prindeville b9df2c481b test: Add ability to capture child process output
ASTERISK-30037

Change-Id: Icbf84ce05addb197a458361c35d784e460d8d6c2
2022-09-12 08:15:27 -05:00
Philip Prindeville d13afaf302 res_crypto: Don't load non-regular files in keys directory
ASTERISK-30046

Change-Id: Ie77e0648f8b0b1c2159fb24662d1989cfd4cc36d
2022-09-12 07:55:33 -05:00
Naveen Albert 2dac2bf8dc func_frame_trace: Remove bogus assertion.
The FRAME_TRACE function currently asserts if it sees
a MASQUERADE_NOTIFY. However, this is a legitimate thing
that can happen so asserting is inappropriate, as there
are no clear negative ramifications of such a thing. This
is adjusted to be like the other frames to print out
the subclass.

ASTERISK-30210 #close

Change-Id: I8ecbdcf17e35f64bdeab42868471f581ad1d1a56
2022-09-11 18:04:59 -05:00
Naveen Albert c487425620 lock.c: Add AMI event for deadlocks.
Adds an AMI event to indicate that a deadlock
has likely started, when Asterisk is compiled
with DETECT_DEADLOCKS enabled. This can make
it easier to perform automated deadlock detection
and take appropriate action (such as doing a core
dump). Unlike the deadlock warnings, the AMI event
is emitted only once per deadlock.

ASTERISK-30161 #close

Change-Id: Ifc6ed3e390f8b4cff7f8077a50e4d7a5b54e42fb
2022-09-11 18:02:24 -05:00
Naveen Albert 205c7c8d21 app_confbridge: Add end_marked_any option.
Adds the end_marked_any option, which can be used
to kick a user from a conference if any marked user
leaves.

ASTERISK-30211 #close

Change-Id: I9e8da7ccb892e522546c0f2b5476d172e022c2f5
2022-09-11 16:22:18 -05:00
Naveen Albert 2de016b181 pbx_variables: Use const char if possible.
Use const char for char arguments to
pbx_substitute_variables_helper_full_location
that can do so (context and exten).

ASTERISK-30209 #close

Change-Id: I001357177e9c3dca2b2b4eebc5650c1095b3da6f
2022-09-11 08:33:06 -05:00
George Joseph 05f42806cc res_geolocation: Add two new options to GEOLOC_PROFILE
Added an 'a' option to the GEOLOC_PROFILE function to allow
variable lists like location_info_refinement to be appended
to instead of replacing the entire list.

Added an 'r' option to the GEOLOC_PROFILE function to resolve all
variables before a read operation and after a Set operation.

Added a few missing parameters to the ones allowed for writing
with GEOLOC_PROFILE.

Fixed a bug where calling GEOLOC_PROFILE to read a parameter
might actually update the profile object.

Cleaned up XML documentation a bit.

ASTERISK-30190

Change-Id: I75f541db43345509a2e86225bfa4cf8e242e5b6c
2022-09-10 12:54:24 -05:00
George Joseph c799db6a21 res_geolocation: Allow location parameters on the profile object
You can now specify the location object's format, location_info,
method, location_source and confidence parameters directly on
a profile object for simple scenarios where the location
information isn't common with any other profiles.  This is
mutually exclusive with setting location_reference on the
profile.

Updated appdocsxml.dtd to allow xi:include in a configObject
element.  This makes it easier to link to complete configOptions
in another object.  This is used to add the above fields to the
profile object without having to maintain the option descriptions
in two places.

ASTERISK-30185

Change-Id: Ifd5f05be0a76f0a6ad49fa28d17c394027677569
2022-09-10 12:51:02 -05:00
George Joseph 4ffc5561c4 res_geolocation: Add profile parameter suppress_empty_ca_elements
Added profile parameter "suppress_empty_ca_elements" that
will cause Civic Address elements that are empty to be
suppressed from the outgoing PIDF-LO document.

Fixed a possible SEGV if a sub-parameter value didn't have a
value.

ASTERISK-30177

Change-Id: I924ccc5aa2f45110a3155b22e53dfaf3ef2092dd
2022-09-10 11:07:51 -05:00
George Joseph 2d5a6498dd res_geolocation: Add built-in profiles
The trigger to perform outgoing geolocation processing is the
presence of a geoloc_outgoing_call_profile on an endpoint. This
is intentional so as to not leak location information to
destinations that shouldn't receive it.   In a totally dynamic
configuration scenario however, there may not be any profiles
defined in geolocation.conf.  This makes it impossible to do
outgoing processing without defining a "dummy" profile in the
config file.

This commit adds 4 built-in profiles:
  "<prefer_config>"
  "<discard_config>"
  "<prefer_incoming>"
  "<discard_incoming>"
The profiles are empty except for having their precedence
set and can be set on an endpoint to allow processing without
entries in geolocation.conf.  "<discard_config>" is actually the
best one to use in this situation.

ASTERISK-30182

Change-Id: I1819ccfa404ce59802a3a07ad1cabed60fb9480a
2022-09-10 11:04:46 -05:00
Joshua C. Colp f3de933b16 res_pjsip_sdp_rtp: Skip formats without SDP details.
When producing an outgoing SDP we iterate through the configured
formats and produce SDP information. It is possible for some
configured formats to not have SDP information available. If this
is the case we skip over them to allow the SDP to still be
produced.

ASTERISK-29185

Change-Id: I3e37569aa4ca341260e6ca5904dc2f75e46a1749
2022-09-10 11:00:38 -05:00
Naveen Albert c7612521be cli: Prevent assertions on startup from bad ao2 refs.
If "core show channels" is run before startup has completed, it
is possible for bad ao2 refs to occur because the system is not
yet fully initialized. This will lead to an assertion failing.

To prevent this, initialization of CLI builtins is moved to be
later along in the main load sequence. Core CLI commands are
loaded at the same time, but channel-related commands are loaded
later on.

ASTERISK-29846 #close

Change-Id: If6b3cde802876bd738c1b4cf2683bea6ddc615b6
2022-09-09 20:41:46 -05:00
Joshua C. Colp a0713a9f70 pjsip: Add TLS transport reload support for certificate and key.
This change adds support using the pjsip_tls_transport_restart
function for reloading the TLS certificate and key, if the filenames
remain unchanged. This is useful for Let's Encrypt and other
situations. Note that no restart of the transport will occur if
the certificate and key remain unchanged.

ASTERISK-30186

Change-Id: I9bc95a6bf791830a9491ad9fa43c17d4010028d0
2022-09-09 18:41:12 -05:00
Naveen Albert 754346a4a9 res_tonedetect: Fix typos referring to wrong variables.
Fixes two typos that cause fax detection to not work.
One refers to the wrong frame variable, and the other
refers to the subclass.integer instead of the frametype
as it should.

ASTERISK-30192 #close

Change-Id: I7b35fdb7bcf25a29a212eee37c20812c64ab3ef1
2022-09-09 13:31:39 -05:00
Mike Bradeen 46776c77c4 alembic: add missing ps_endpoints columns
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
2022-09-09 11:35:04 -05:00
Sean Bright 583e017f34 chan_dahdi.c: Resolve a format-truncation build warning.
With gcc (Ubuntu 11.2.0-19ubuntu1) 11.2.0:

> chan_dahdi.c:4129:18: error: ‘%s’ directive output may be truncated
>   writing up to 255 bytes into a region of size between 242 and 252
>   [-Werror=format-truncation=]

This removes the error-prone sizeof(...) calculations in favor of just
doubling the size of the base buffer.

Change-Id: I2d276785286730d3d5d0a921bcea2e065dbf27c5
2022-09-09 09:39:21 -05:00
Alexei Gradinari 12c4c1bf5f res_pjsip_pubsub: Postpone destruction of old subscriptions on RLS update
Set termination state to old subscriptions to prevent queueing and sending
NOTIFY messages on exten/device state changes.

Postpone destruction of old subscriptions until all already queued tasks
that may be using old subscriptions have completed.

ASTERISK-29906

Change-Id: I96582aad3a26515ca73a8460ee6756f56f6ba23b
2022-09-09 08:35:53 -05:00
Sean Bright 155c796203 channel.h: Remove redundant declaration.
The DECLARE_STRINGFIELD_SETTERS_FOR() declares ast_channel_name_set()
for us, so no need to declare it separately.

Change-Id: I4813a884ada475ddc62bca480bceb4a53b3ec59a
2022-09-09 05:59:45 -05:00
Naveen Albert 3fa66c92b5 features: Add transfer initiation options.
Adds additional control options over the transfer
feature functionality to give users more control
in how the transfer feature sounds and works.

First, the "transfer" sound that plays when a transfer is
initiated can now be customized by the user in
features.conf, just as with the other transfer sounds.

Secondly, the user can now specify the transfer extension
in advance by using the TRANSFER_EXTEN variable. If
a valid extension is contained in this variable, the call
will automatically be transferred to this destination.
Otherwise, it will fall back to collecting the extension
from the user as is always done now.

ASTERISK-29899 #close

Change-Id: Ibff309caa459a2b958706f2ed0ca393b1ef502e3
2022-09-08 13:47:25 -05:00
Mike Bradeen adffb975dc CI: Fixing path issue on venv check
ASTERISK-26826

Change-Id: I07388d16f74452cebc9c981f99044eb6b77df792
2022-08-31 15:56:52 -05:00
Mike Bradeen 4fc9e06db1 CI: use Python3 virtual environment
Requires Python3 testsuite changes

ASTERISK-26826

Change-Id: I92ec7dec751ad455503a584d6e860db88c56d6bc
2022-08-30 11:41:54 -05:00
Naveen Albert e2e049e473 general: Very minor coding guideline fixes.
Fixes a few coding guideline violations:
* Use of C99 comments
* Opening brace on same line as function prototype

ASTERISK-30163 #close

Change-Id: I07771c4c89facd41ce8d323859f022ddbddf6ca7
2022-08-17 11:05:08 -05:00
George Joseph 8a8416e365 res_geolocation: Address user issues, remove complexity, plug leaks
* Added processing for the 'confidence' element.
* Added documentation to some APIs.
* removed a lot of complex code related to the very-off-nominal
  case of needing to process multiple location info sources.
* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
  one eprofile instead of a datastore of multiples.
* Plugged a huge leak in XML processing that arose from
  insufficient documentation by the libxml/libxslt authors.
* Refactored stylesheets to be more efficient.
* Renamed 'profile_action' to 'profile_precedence' to better
  reflect it's purpose.
* Added the config option for 'allow_routing_use' which
  sets the value of the 'Geolocation-Routing' header.
* Removed the GeolocProfileCreate and GeolocProfileDelete
  dialplan apps.
* Changed the GEOLOC_PROFILE dialplan function as follows:
  * Removed the 'profile' argument.
  * Automatically create a profile if it doesn't exist.
  * Delete a profile if 'inheritable' is set to no.
* Fixed various bugs and leaks
* Updated Asterisk WiKi documentation.

ASTERISK-30167

Change-Id: If38c23f26228e96165be161c2f5e849cb8e16fa0
2022-08-10 12:50:01 -05:00
Naveen Albert ff044c222b chan_iax2: Add missing options documentation.
Adds missing dial resource option documentation.

ASTERISK-30164 #close

Change-Id: I674e1fc9b1e5d67a20599bd4b418ce294d48fc83
2022-08-10 08:36:21 -05:00
Naveen Albert dc7ec11c26 app_confbridge: Fix memory leak on updated menu options.
If the CONFBRIDGE function is used to dynamically set
menu options, a memory leak occurs when a menu option
that has been set is overridden, since the menu entry
is not destroyed before being freed. This ensures that
it is.

Additionally, logic that duplicates the destroy function
is removed in lieu of the destroy function itself.

ASTERISK-28422 #close

Change-Id: I71cfb5c24e636984d41086d1333a416dc12ff995
2022-08-08 05:19:40 -05:00
George Joseph 30d7a212b0 Geolocation: Wiki Documentation
Change-Id: I68ba22db0a69d9e2eabcc2141b48a2395f7f1a23
2022-08-05 10:02:30 -05:00
Naveen Albert f4a020a45b manager: Remove documentation for nonexistent action.
The manager XML documentation documents a "FilterList"
action, but there is no such action. Therefore, this can
lead to confusion when people try to use a documented
action that does not, in fact, exist. This is removed
as the action never did exist in the past, nor would it
be trivial to add since we only store the regex_t
objects, so the filter list can't actually be provided
without storing that separately. Most likely, the
documentation was originally added (around version 10)
in anticipation of something that never happened.

ASTERISK-29917 #close

Change-Id: I846b16fd6f80a91d4ddc5d8a861b522d7c6f8f97
2022-08-02 09:09:43 -05:00
Naveen Albert c654486547 general: Improve logging levels of some log messages.
Adjusts some logging levels to be more or less important,
that is more prominent when actual problems occur and less
prominent for less noteworthy things.

ASTERISK-30153 #close

Change-Id: Ifc8f7df427aa018627db462125ae744986d3261b
2022-08-01 11:03:46 -05:00
Naveen Albert 5feebc0857 cdr.conf: Remove obsolete app_mysql reference.
The CDR sample config still mentions that app_mysql
is available in the addons directory, but this is
incorrect as it was removed as of 19. This removes
that to avoid confusion.

ASTERISK-30160 #close

Change-Id: Ie5293ccb4f2b365896981811b480544e67bb9cd7
2022-08-01 10:09:53 -05:00
Naveen Albert 165368bf0b general: Remove obsolete SVN references.
There are a handful of files in the tree that
reference an SVN link for the coding guidelines.

This removes these because the links are dead
and the vast majority of source files do not
contain these links, so this is more consistent.

app_skel still maintains an (up to date) link
to the coding guidelines.

ASTERISK-30159 #close

Change-Id: I35bbb20f66982e98099cff3029ede20091ffdac7
2022-08-01 09:58:58 -05:00
Naveen Albert 2d8f2696b2 app_confbridge: Add missing AMI documentation.
Documents the ConfbridgeListRooms AMI response,
which is currently not documented.

ASTERISK-30020 #close

Change-Id: Id6fff7a936244bae7b52686301eb740c1169cdea
2022-08-01 08:58:50 -05:00
Naveen Albert 4af881506e app_meetme: Add missing AMI documentation.
The MeetmeList and MeetmeListRooms AMI
responses are currently completely undocumented.
This adds documentation for these responses.

ASTERISK-30018 #close

Change-Id: Id93135b7edf01de6f8fba266e2122989dc8996b8
2022-08-01 07:57:21 -05:00
Naveen Albert 83912496ab func_srv: Document field parameter.
Adds missing documentation for the field parameter
for the SRVRESULT function.

ASTERISK-30151
Reported by: Chris Young

Change-Id: I4385a2e0892a07e30dea1a8a0588e2c1bea2b1f1
2022-07-27 12:10:25 -05:00
Naveen Albert c771e2dd7a pbx_functions.c: Manually update ast_str strlen.
When ast_func_read2 is used to read a function using
its read function (as opposed to a native ast_str read2
function), the result is copied directly by the function
into the ast_str buffer. As a result, the ast_str length
remains initialized to 0, which is a bug because this is
not the real string length.

This can cascade and have issues elsewhere, such as when
reading substrings of functions that only register read
as opposed to read2 callbacks. In this case, since reading
ast_str_strlen returns 0, the returned substring is empty
as opposed to the actual substring. This has caused
the ast_str family of functions to behave inconsistently
and erroneously, in contrast to the pbx_variables substitution
functions which work correctly.

This fixes this issue by manually updating the ast_str length
when the result is copied directly into the ast_str buffer.

Additionally, an assertion and a unit test that previously
exposed these issues are added, now that the issue is fixed.

ASTERISK-29966 #close

Change-Id: I4e2dba41410f9d4dff61c995d2ca27718248e07f
2022-07-26 10:48:19 -05:00
Sergey V. Lobanov f645157a4b build: fix bininstall launchd issue on cross-platform build
configure script detects /sbin/launchd, but the result of this
check is not used in Makefile (bininstall). Makefile also detects
/sbin/launchd file to decide if it is required to install
safe_asterisk.

configure script correctly detects cross compile build and sets
PBX_LAUNCHD=0

In case of building asterisk on MacOS host for Linux target using
external toolchain (e.g. OpenWrt toolchain), bininstall does not
install safe_asterisk (due to /sbin/launchd detection in Makefile),
but it is required on target (Linux).

This patch adds HAVE_SBIN_LAUNCHD=@PBX_LAUNCHD@ to makeopts.in to
use the result of /sbin/launchd detection from configure script in
Makefile.
Also this patch uses HAVE_SBIN_LAUNCHD in Makefile (bininstall) to
decide if it is required to install safe_asterisk.

ASTERISK-29905 #close

Change-Id: Iff61217276cd188f43f51ef4cdbffe39d9f07f65
2022-07-24 11:52:30 -05:00
Naveen Albert a9223f210e db: Add AMI action to retrieve DB keys at prefix.
Adds the DBGetTree action, which can be used to
retrieve all of the DB keys beginning with a
particular prefix, similar to the capability
provided by the database show CLI command.

ASTERISK-30136 #close

Change-Id: I3be9425e53be71f24303fdd4d2923c14e84337e6
2022-07-20 13:02:12 -05:00
Naveen Albert ce18196280 manager: Fix incomplete filtering of AMI events.
The global event filtering code was only in one
possible execution path, so not all events were
being properly filtered out if requested. This moves
that into the universal AMI handling code so all
events are properly handled.

Additionally, the CLI listing of disabled events can
also get truncated, so we now print out everything.

ASTERISK-30137 #close

Change-Id: If8c42edcb2abc5158552da7eba2a8ff6b20e1959
2022-07-20 09:46:11 -05:00
George Joseph f8000daff5 Update defaultbranch to 20
Change-Id: Ib91db9223a78188667e15053bcc73931b878414e
2022-07-20 04:59:53 -06:00
820 changed files with 192989 additions and 51731 deletions

87
.github/ISSUE_TEMPLATE/bug-report.yml vendored Normal file
View File

@ -0,0 +1,87 @@
name: Bug
description: File a bug report
title: "[bug]: "
labels: ["bug", "triage"]
#assignees:
# - octocat
body:
- type: markdown
attributes:
value: |
Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.
A good first step is for you to review the Asterisk Issue Guidelines if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
Then, if you are submitting a patch, please review the Patch Contribution Process.
Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
Please note that by submitting data, code, or documentation to Sangoma through GitHub, you accept the Terms of Use present at
https://www.asterisk.org/terms-of-use/.
Thanks for taking the time to fill out this bug report!
- type: dropdown
id: severity
attributes:
label: Severity
options:
- Trivial
- Minor
- Major
- Critical
- Blocker
validations:
required: true
- type: input
id: versions
attributes:
label: Versions
description: Enter one or more versions separated by commas.
validations:
required: true
- type: input
id: components
attributes:
label: Components/Modules
description: Enter one or more components or modules separated by commas.
validations:
required: true
- type: textarea
id: environment
attributes:
label: Operating Environment
description: OS, Disribution, Version, etc.
validations:
required: true
- type: dropdown
id: frequency
attributes:
label: Frequency of Occurrence
options:
- "Never"
- "One Time"
- "Occasional"
- "Frequent"
- "Constant"
- type: textarea
id: description
attributes:
label: Issue Description
validations:
required: true
- type: textarea
id: logs
attributes:
label: Relevant log output
description: Please copy and paste any relevant log output. This will be automatically formatted into code, so no need for backticks.
render: shell
- type: markdown
attributes:
value: |
[Asterisk Issue Guidelines](https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines)
- type: checkboxes
id: guidelines
attributes:
label: Asterisk Issue Guidelines
options:
- label: Yes, I have read the Asterisk Issue Guidelines
required: true

11
.github/ISSUE_TEMPLATE/config.yml vendored Normal file
View File

@ -0,0 +1,11 @@
blank_issues_enabled: false
contact_links:
- name: Asterisk Community Support
url: https://community.asterisk.org
about: Please ask and answer questions here.
- name: Feature Requests (Without Code)
url: https://github.com/asterisk/asterisk-feature-requests/issues
about: Please submit feature requests (without code) here.
- name: Improvement Requests (Without Code)
url: https://github.com/asterisk/asterisk-feature-requests/issues
about: Please submit improvement requests (without code) here.

27
.github/ISSUE_TEMPLATE/improvement.yml vendored Normal file
View File

@ -0,0 +1,27 @@
name: Improvement
description: Submit an improvement to existing functionality
title: "[improvement]: "
labels: ["improvement", "triage"]
body:
- type: markdown
attributes:
value: |
Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.
A good first step is for you to review the Asterisk Issue Guidelines if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
Then, if you are submitting a patch, please review the Patch Contribution Process.
Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
Please note that by submitting data, code, or documentation to Sangoma through GitHub, you accept the Terms of Use present at
https://www.asterisk.org/terms-of-use/.
Thanks for taking the time to fill out this bug report!
- type: textarea
id: description
attributes:
label: Improvement Description
description: Describe the improvement in as much detail as possible
validations:
required: true

27
.github/ISSUE_TEMPLATE/new-feature.yml vendored Normal file
View File

@ -0,0 +1,27 @@
name: New Feature Submission
description: Submit a New Feature
title: "[new-feature]: "
labels: ["new-feature", "triage"]
body:
- type: markdown
attributes:
value: |
Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.
A good first step is for you to review the Asterisk Issue Guidelines if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
Then, if you are submitting a patch, please review the Patch Contribution Process.
Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
Please note that by submitting data, code, or documentation to Sangoma through GitHub, you accept the Terms of Use present at
https://www.asterisk.org/terms-of-use/.
Thanks for taking the time to fill out this bug report!
- type: textarea
id: description
attributes:
label: Feature Description
description: Describe the new feature in as much detail as possible
validations:
required: true

167
.github/workflows/CherryPickTest.yml vendored Normal file
View File

@ -0,0 +1,167 @@
name: CherryPickTest
run-name: "Cherry-Pick Tests for PR ${{github.event.number}}"
on:
pull_request_target:
types: [ labeled ]
concurrency:
group: ${{github.workflow}}-${{github.event.number}}
cancel-in-progress: true
env:
PR_NUMBER: ${{ github.event.number }}
MODULES_BLACKLIST: ${{ vars.GATETEST_MODULES_BLACKLIST }} ${{ vars.UNITTEST_MODULES_BLACKLIST }}
jobs:
IdentifyBranches:
name: IdentifyBranches
if: ${{ github.event.label.name == vars.CHERRY_PICK_TEST_LABEL }}
outputs:
branches: ${{ steps.getbranches.outputs.branches }}
branch_count: ${{ steps.getbranches.outputs.branch_count }}
runs-on: ubuntu-latest
steps:
- name: Remove Trigger Label, Add InProgress Label
env:
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
run: |
gh pr edit --repo ${{github.repository}} \
--remove-label ${{vars.CHERRY_PICK_TEST_LABEL}} \
--remove-label ${{vars.CHERRY_PICK_CHECKS_PASSED_LABEL}} \
--remove-label ${{vars.CHERRY_PICK_CHECKS_FAILED_LABEL}} \
--remove-label ${{vars.CHERRY_PICK_GATES_PASSED_LABEL}} \
--remove-label ${{vars.CHERRY_PICK_GATES_FAILED_LABEL}} \
--remove-label ${{vars.CHERRY_PICK_TESTING_IN_PROGRESS}} \
${{env.PR_NUMBER}} || :
- name: Get cherry-pick branches
uses: asterisk/asterisk-ci-actions/GetCherryPickBranchesFromPR@main
id: getbranches
with:
repo: ${{github.repository}}
pr_number: ${{env.PR_NUMBER}}
cherry_pick_regex: ${{vars.CHERRY_PICK_REGEX}}
github_token: ${{secrets.GITHUB_TOKEN}}
- name: Check Branch Count
if: ${{ steps.getbranches.outputs.branch_count > 0 }}
env:
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
run: |
gh pr edit --repo ${{github.repository}} \
--add-label ${{vars.CHERRY_PICK_TESTING_IN_PROGRESS}} \
${{env.PR_NUMBER}} || :
CherryPickUnitTestMatrix:
needs: [ IdentifyBranches ]
if: ${{ needs.IdentifyBranches.outputs.branch_count > 0 && ( success() || failure() ) }}
continue-on-error: false
strategy:
fail-fast: false
matrix:
branch: ${{ fromJSON(needs.IdentifyBranches.outputs.branches) }}
runs-on: ubuntu-latest
steps:
- name: Run Unit Tests for branch ${{matrix.branch}}
uses: asterisk/asterisk-ci-actions/AsteriskUnitComposite@main
with:
asterisk_repo: ${{github.repository}}
pr_number: ${{env.PR_NUMBER}}
base_branch: ${{matrix.branch}}
is_cherry_pick: true
modules_blacklist: ${{env.MODULES_BLACKLIST}}
github_token: ${{secrets.GITHUB_TOKEN}}
unittest_command: ${{vars.UNITTEST_COMMAND}}
CherryPickUnitTests:
needs: [ IdentifyBranches, CherryPickUnitTestMatrix ]
if: ${{ needs.IdentifyBranches.outputs.branch_count > 0 && ( success() || failure() ) }}
runs-on: ubuntu-latest
steps:
- name: Check unit test matrix status
env:
RESULT: ${{needs.CherryPickUnitTestMatrix.result}}
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
run: |
case $RESULT in
success)
gh pr edit --repo ${{github.repository}} \
--add-label ${{vars.CHERRY_PICK_CHECKS_PASSED_LABEL}} \
${{env.PR_NUMBER}} || :
echo "::notice::All tests passed"
exit 0
;;
skipped)
gh pr edit --repo ${{github.repository}} \
--remove-label ${{vars.CHERRY_PICK_TESTING_IN_PROGRESS}} \
--add-label ${{vars.CHERRY_PICK_CHECKS_FAILED_LABEL}} \
${{env.PR_NUMBER}} || :
echo "::notice::Unit tests were skipped because of an earlier failure"
exit 1
;;
*)
gh pr edit --repo ${{github.repository}} \
--remove-label ${{vars.CHERRY_PICK_TESTING_IN_PROGRESS}} \
--add-label ${{vars.CHERRY_PICK_CHECKS_FAILED_LABEL}} \
${{env.PR_NUMBER}} || :
echo "::error::One or more tests failed ($RESULT)"
exit 1
esac
CherryPickGateTestMatrix:
needs: [ IdentifyBranches, CherryPickUnitTests ]
if: ${{ success() }}
continue-on-error: false
strategy:
fail-fast: false
matrix:
branch: ${{ fromJSON(needs.IdentifyBranches.outputs.branches) }}
group: ${{ fromJSON(vars.GATETEST_LIST) }}
runs-on: ubuntu-latest
steps:
- name: Run Gate Tests for ${{ matrix.group }}-${{matrix.branch}}
uses: asterisk/asterisk-ci-actions/AsteriskGateComposite@main
with:
test_type: Gate
asterisk_repo: ${{github.repository}}
pr_number: ${{env.PR_NUMBER}}
base_branch: ${{matrix.branch}}
is_cherry_pick: true
modules_blacklist: ${{env.MODULES_BLACKLIST}}
github_token: ${{secrets.GITHUB_TOKEN}}
testsuite_repo: ${{vars.TESTSUITE_REPO}}
gatetest_group: ${{matrix.group}}
gatetest_command: ${{ toJSON(fromJSON(vars.GATETEST_COMMANDS)[matrix.group]) }}
CherryPickGateTests:
needs: [ IdentifyBranches, CherryPickGateTestMatrix ]
if: ${{ success() || failure() }}
runs-on: ubuntu-latest
steps:
- name: Check test matrix status
env:
RESULT: ${{needs.CherryPickGateTestMatrix.result}}
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
run: |
gh pr edit --repo ${{github.repository}} \
--remove-label ${{vars.CHERRY_PICK_TESTING_IN_PROGRESS}} \
${{env.PR_NUMBER}} || :
case $RESULT in
success)
gh pr edit --repo ${{github.repository}} \
--add-label ${{vars.CHERRY_PICK_GATES_PASSED_LABEL}} \
${{env.PR_NUMBER}} || :
echo "::notice::All Testsuite tests passed"
exit 0
;;
skipped)
echo "::error::Testsuite tests were skipped because of an earlier failure"
exit 1
;;
*)
gh pr edit --repo ${{github.repository}} \
--add-label ${{vars.CHERRY_PICK_GATES_FAILED_LABEL}} \
${{env.PR_NUMBER}} || :
echo "::error::One or more Testsuite tests failed ($RESULT)"
exit 1
esac

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name: CreateDocs
on:
workflow_dispatch:
inputs:
branches:
description: "JSON array of branches: ['18','20'] (no spaces)"
required: false
type: string
schedule:
# Times are UTC
- cron: '0 04 * * *'
env:
ASTERISK_REPO: ${{ github.repository }}
GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
DEFAULT_BRANCHES: ${{ vars.WIKIDOC_BRANCHES }}
INPUT_BRANCHES: ${{ inputs.branches }}
jobs:
CreateDocsDebug:
runs-on: ubuntu-latest
outputs:
manual_branches: ${{ steps.setup.outputs.manual_branches }}
steps:
- name: setup
run: |
MANUAL_BRANCHES="$INPUT_BRANCHES"
[ -z "$MANUAL_BRANCHES" ] && MANUAL_BRANCHES="$DEFAULT_BRANCHES" || :
echo "manual_branches=${MANUAL_BRANCHES}"
echo "manual_branches=${MANUAL_BRANCHES}" >>${GITHUB_OUTPUT}
exit 0
- name: DumpEnvironment
uses: asterisk/asterisk-ci-actions/DumpEnvironmentAction@main
with:
action-inputs: ${{toJSON(inputs)}}
action-vars: ${{ toJSON(steps.setup.outputs) }}
CreateDocsScheduledMatrix:
needs: [ CreateDocsDebug ]
if: ${{github.event_name == 'schedule' && fromJSON(vars.WIKIDOCS_ENABLE) == true }}
continue-on-error: false
strategy:
fail-fast: false
matrix:
branch: ${{ fromJSON(vars.WIKIDOC_BRANCHES) }}
runs-on: ubuntu-latest
steps:
- name: CreateDocs for ${{matrix.branch}}
uses: asterisk/asterisk-ci-actions/CreateAsteriskDocsComposite@main
with:
asterisk_repo: ${{env.ASTERISK_REPO}}
base_branch: ${{matrix.branch}}
docs_dir: docs_dir/${{matrix.branch}}
github_token: ${{secrets.GITHUB_TOKEN}}
CreateDocsScheduled:
needs: [ CreateDocsScheduledMatrix ]
if: ${{ success() || failure() }}
runs-on: ubuntu-latest
steps:
- name: Check CreateDocsScheduledMatrix status
env:
RESULT: ${{needs.CreateDocsScheduledMatrix.result}}
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
run: |
case $RESULT in
success)
echo "::notice::Docs created"
exit 0
;;
skipped)
echo "::notice::Skipped"
exit 1
;;
*)
echo "::error::One or CreateDocs failed ($RESULT)"
exit 1
esac
CreateDocsManualMatrix:
needs: [ CreateDocsDebug ]
if: ${{github.event_name == 'workflow_dispatch'}}
continue-on-error: false
strategy:
fail-fast: false
matrix:
branch: ${{ fromJSON(vars.WIKIDOC_MANUAL_BRANCHES) }}
runs-on: ubuntu-latest
steps:
- name: CreateDocs for ${{matrix.branch}}
uses: asterisk/asterisk-ci-actions/CreateAsteriskDocsComposite@main
with:
asterisk_repo: ${{env.ASTERISK_REPO}}
base_branch: ${{matrix.branch}}
docs_dir: docs_dir/${{matrix.branch}}
github_token: ${{secrets.GITHUB_TOKEN}}
CreateDocsManual:
needs: [ CreateDocsManualMatrix ]
if: ${{ success() || failure() }}
runs-on: ubuntu-latest
steps:
- name: Check CreateDocsManualMatrix status
env:
RESULT: ${{needs.CreateDocsManualMatrix.result}}
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
run: |
case $RESULT in
success)
echo "::notice::Docs created"
exit 0
;;
skipped)
echo "::notice::Skipped"
exit 1
;;
*)
echo "::error::One or CreateDocs failed ($RESULT)"
exit 1
esac

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name: Issue Opened
run-name: "Issue ${{github.event.number}} ${{github.event.action}} by ${{github.actor}}"
on:
issues:
types: opened
jobs:
triage:
runs-on: ubuntu-latest
steps:
- name: initial labeling
uses: andymckay/labeler@master
with:
add-labels: "triage"
ignore-if-labeled: true

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name: MergeApproved
run-name: "Merge Approved for PR ${{github.event.number}}"
on:
pull_request_target:
types: [labeled]
env:
PR_NUMBER: ${{ github.event.number }}
BASE_BRANCH: ${{github.event.pull_request.base.ref}}
MODULES_BLACKLIST: ${{ vars.GATETEST_MODULES_BLACKLIST }} ${{ vars.UNITTEST_MODULES_BLACKLIST }}
FORCE: ${{ endsWith(github.event.label.name, '-force') }}
jobs:
IdentifyBranches:
if: contains(fromJSON(vars.MERGE_APPROVED_LABELS), github.event.label.name)
outputs:
branches: ${{ steps.getbranches.outputs.branches }}
all_branches: ${{ steps.checkbranches.outputs.all_branches }}
branch_count: ${{ steps.getbranches.outputs.branch_count }}
runs-on: ubuntu-latest
steps:
- name: Clean up labels
env:
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
run: |
gh pr edit --repo ${{github.repository}} \
--remove-label ${{github.event.label.name}} \
--remove-label ${{vars.PRE_MERGE_CHECKS_PASSED_LABEL}} \
--remove-label ${{vars.PRE_MERGE_CHECKS_FAILED_LABEL}} \
--remove-label ${{vars.PRE_MERGE_GATES_PASSED_LABEL}} \
--remove-label ${{vars.PRE_MERGE_GATES_FAILED_LABEL}} \
--remove-label ${{vars.PRE_MERGE_TESTING_IN_PROGRESS}} \
${{env.PR_NUMBER}} || :
- name: Get cherry-pick branches
uses: asterisk/asterisk-ci-actions/GetCherryPickBranchesFromPR@main
id: getbranches
with:
repo: ${{github.repository}}
pr_number: ${{env.PR_NUMBER}}
cherry_pick_regex: ${{vars.CHERRY_PICK_REGEX}}
github_token: ${{secrets.GITHUB_TOKEN}}
- name: Check Branch Count
id: checkbranches
env:
BRANCH_COUNT: ${{ steps.getbranches.outputs.branch_count }}
BRANCHES: ${{ steps.getbranches.outputs.branches }}
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
run: |
gh pr edit --repo ${{github.repository}} \
--add-label ${{vars.PRE_MERGE_TESTING_IN_PROGRESS}} \
${{env.PR_NUMBER}} || :
all_branches=$(echo "$BRANCHES" | jq -c "[ \"$BASE_BRANCH\" ] + .")
echo "all_branches=${all_branches}" >>${GITHUB_OUTPUT}
- name: Pre Check Cherry-Picks
if: ${{ steps.getbranches.outputs.branch_count > 0 }}
uses: asterisk/asterisk-ci-actions/CherryPick@main
with:
repo: ${{github.repository}}
pr_number: ${{env.PR_NUMBER}}
branches: ${{steps.getbranches.outputs.branches}}
github_token: ${{secrets.GITHUB_TOKEN}}
push: false
PreMergeUnitTestMatrix:
needs: [ IdentifyBranches ]
if: success()
continue-on-error: false
strategy:
fail-fast: false
matrix:
branch: ${{ fromJSON(needs.IdentifyBranches.outputs.all_branches) }}
runs-on: ubuntu-latest
steps:
- name: Run Unit Tests for branch ${{matrix.branch}}
uses: asterisk/asterisk-ci-actions/AsteriskUnitComposite@main
with:
asterisk_repo: ${{github.repository}}
pr_number: ${{env.PR_NUMBER}}
base_branch: ${{matrix.branch}}
is_cherry_pick: true
modules_blacklist: ${{env.MODULES_BLACKLIST}}
github_token: ${{secrets.GITHUB_TOKEN}}
unittest_command: ${{vars.UNITTEST_COMMAND}}
PreMergeUnitTests:
needs: [ IdentifyBranches, PreMergeUnitTestMatrix ]
runs-on: ubuntu-latest
steps:
- name: Check unit test matrix status
env:
RESULT: ${{needs.PreMergeUnitTestMatrix.result}}
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
run: |
case $RESULT in
success)
gh pr edit --repo ${{github.repository}} \
--remove-label ${{vars.PRE_MERGE_TESTING_IN_PROGRESS}} \
--add-label ${{vars.PRE_MERGE_CHECKS_PASSED_LABEL}} \
${{env.PR_NUMBER}} || :
echo "::notice::All tests passed"
exit 0
;;
skipped)
gh pr edit --repo ${{github.repository}} \
--remove-label ${{vars.PRE_MERGE_TESTING_IN_PROGRESS}} \
--add-label ${{vars.PRE_MERGE_CHECKS_FAILED_LABEL}} \
${{env.PR_NUMBER}} || :
echo "::notice::Unit tests were skipped because of an earlier failure"
exit 1
;;
*)
gh pr edit --repo ${{github.repository}} \
--remove-label ${{vars.PRE_MERGE_TESTING_IN_PROGRESS}} \
--add-label ${{vars.PRE_MERGE_CHECKS_FAILED_LABEL}} \
${{env.PR_NUMBER}} || :
echo "::error::One or more tests failed ($RESULT)"
exit 1
esac
MergeAndCherryPick:
needs: [ IdentifyBranches, PreMergeUnitTests ]
if: success()
runs-on: ubuntu-latest
steps:
- name: Start Merge
env:
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
run: |
gh pr edit --repo ${{github.repository}} \
--add-label ${{vars.MERGE_IN_PROGRESS_LABEL}} \
${{env.PR_NUMBER}} || :
- name: Get Token needed to push cherry-picks
id: get_workflow_token
uses: peter-murray/workflow-application-token-action@v2
with:
application_id: ${{secrets.ASTERISK_ORG_ACCESS_APP_ID}}
application_private_key: ${{secrets.ASTERISK_ORG_ACCESS_APP_PRIV_KEY}}
organization: asterisk
- name: Merge and Cherry Pick to ${{needs.IdentifyBranches.outputs.branches}}
id: mergecp
uses: asterisk/asterisk-ci-actions/MergeAndCherryPickComposite@main
with:
repo: ${{github.repository}}
pr_number: ${{env.PR_NUMBER}}
branches: ${{needs.IdentifyBranches.outputs.branches}}
force: ${{env.FORCE}}
github_token: ${{steps.get_workflow_token.outputs.token}}
- name: Merge Cleanup
if: always()
env:
RESULT: ${{ steps.mergecp.outcome }}
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
BRANCH_COUNT: ${{ needs.IdentifyBranches.outputs.branch_count }}
BRANCHES: ${{ needs.IdentifyBranches.outputs.branches }}
run: |
case $RESULT in
success)
gh pr edit --repo ${{github.repository}} \
--remove-label ${{vars.MERGE_IN_PROGRESS_LABEL}} \
${{env.PR_NUMBER}} || :
if [ $BRANCH_COUNT -eq 0 ] ; then
gh pr comment --repo ${{github.repository}} \
-b "Successfully merged to branch $BASE_BRANCH." \
${{env.PR_NUMBER}} || :
else
gh pr comment --repo ${{github.repository}} \
-b "Successfully merged to branch $BASE_BRANCH and cherry-picked to $BRANCHES" \
${{env.PR_NUMBER}} || :
fi
exit 0
;;
failure)
gh pr edit --repo ${{github.repository}} \
--remove-label ${{vars.MERGE_IN_PROGRESS_LABEL}} \
--add-label ${{vars.MERGE_FAILED_LABEL}} \
${{env.PR_NUMBER}} || :
exit 1
;;
*)
esac

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name: Nightly Admin
on:
schedule:
- cron: '30 1 * * *'
env:
ASTERISK_REPO: ${{ github.repository }}
PR_NUMBER: 0
PR_COMMIT: ''
GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
MODULES_BLACKLIST: ${{ vars.GATETEST_MODULES_BLACKLIST }} ${{ vars.UNITTEST_MODULES_BLACKLIST }}
jobs:
CloseStaleIssues:
runs-on: ubuntu-latest
steps:
- name: Close Stale Issues
uses: actions/stale@v7
with:
stale-issue-message: 'This issue is stale because it has been open 7 days with no activity. Remove stale label or comment or this will be closed in 14 days.'
stale-issue-label: stale
close-issue-message: 'This issue was closed because it has been stalled for 14 days with no activity.'
days-before-stale: 7
days-before-close: 14
days-before-pr-close: -1
only-labels: triage,feedback-required

59
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name: NightlyTests
on:
workflow_dispatch:
schedule:
- cron: '0 2 * * *'
env:
ASTERISK_REPO: ${{ github.repository }}
PR_NUMBER: 0
PR_COMMIT: ''
GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
MODULES_BLACKLIST: ${{ vars.GATETEST_MODULES_BLACKLIST }}
jobs:
AsteriskNightly:
strategy:
fail-fast: false
matrix:
branch: ${{ fromJSON(vars.NIGHTLYTEST_BRANCHES) }}
group: ${{ fromJSON(vars.NIGHTLYTEST_LIST) }}
runs-on: ubuntu-latest
steps:
- name: Run Nightly Tests for ${{ matrix.group }}/${{ matrix.branch }}
uses: asterisk/asterisk-ci-actions/AsteriskGateComposite@main
with:
test_type: Nightly
asterisk_repo: ${{env.ASTERISK_REPO}}
pr_number: ${{env.PR_NUMBER}}
base_branch: ${{matrix.branch}}
modules_blacklist: ${{env.MODULES_BLACKLIST}}
github_token: ${{secrets.GITHUB_TOKEN}}
testsuite_repo: ${{vars.TESTSUITE_REPO}}
gatetest_group: ${{matrix.group}}
gatetest_command: ${{ toJSON(fromJSON(vars.GATETEST_COMMANDS)[matrix.group]) }}
AsteriskNightlyTests:
if: ${{ always() }}
runs-on: ubuntu-latest
needs: AsteriskNightly
steps:
- name: Check test matrix status
env:
RESULT: ${{needs.AsteriskNightly.result}}
run: |
case $RESULT in
success)
echo "::notice::All Testsuite tests passed"
exit 0
;;
skipped)
echo "::error::Testsuite tests were skipped because of an earlier failure"
exit 1
;;
*)
echo "::error::One or more Testsuite tests failed"
exit 1
esac

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.github/workflows/PRMerged.yml vendored Normal file
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name: PRMerged
run-name: "PR ${{github.event.number || inputs.pr_number}} ${{github.event.action || 'MANUAL POST MERGE'}} by ${{ github.actor }}"
on:
pull_request_target:
types: [closed]
workflow_dispatch:
inputs:
pr_number:
description: 'PR number'
required: true
type: number
concurrency:
group: ${{github.workflow}}-${{github.event.number || inputs.pr_number}}
cancel-in-progress: true
env:
REPO: ${{github.repository}}
PR_NUMBER: ${{github.event.number || inputs.pr_number}}
GITHUB_TOKEN: ${{secrets.GITHUB_TOKEN}}
jobs:
CloseIssues:
if: github.event.pull_request.merged == true
runs-on: ubuntu-latest
steps:
- uses: wow-actions/auto-close-fixed-issues@v1
env:
GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
with:
GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}

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name: PRSubmitActions
run-name: "PRSubmitActions: Test ${{github.event.action}}"
on:
workflow_run:
workflows: [PRSubmitTests]
types:
- requested
- completed
env:
ACTION: ${{ github.event.action }}
CONCLUSION: ${{ github.event.workflow_run.conclusion }}
REPO: ${{ github.repository }}
jobs:
PRSubmitActions:
runs-on: ubuntu-latest
steps:
- name: Get PR Number
id: getpr
uses: actions/github-script@v7
with:
retries: 5
script: |
let search = `repo:${context.repo.owner}/${context.repo.repo} ${context.payload.workflow_run.head_sha}`;
let prs = await github.rest.search.issuesAndPullRequests({
q: search,
});
if (prs.data.total_count == 0) {
core.setFailed(`Unable to get PR for ${context.payload.workflow_run.head_sha}`);
return;
}
let pr_number = prs.data.items[0].number;
core.setOutput('pr_number', pr_number);
return;
- name: Set Label
id: setlabel
uses: actions/github-script@v7
env:
PR_NUMBER: ${{ steps.getpr.outputs.PR_NUMBER }}
LABEL_TIP: ${{ vars.PR_SUBMIT_TESTING_IN_PROGRESS }}
LABEL_PASS: ${{ vars.PR_SUBMIT_TESTS_PASSED }}
LABEL_FAIL: ${{ vars.PR_SUBMIT_TESTS_FAILED }}
with:
retries: 5
script: |
let label;
if (process.env.ACTION === 'requested') {
label = process.env.LABEL_TIP;
} else {
if ( process.env.CONCLUSION === 'success' ) {
label = process.env.LABEL_PASS;
} else {
label = process.env.LABEL_FAIL;
}
}
core.info(`Setting label ${label}`);
github.rest.issues.setLabels({
issue_number: process.env.PR_NUMBER,
owner: context.repo.owner,
repo: context.repo.repo,
labels: [ label ]
});
return;
- name: Get cherry-pick branches
if: github.event.action == 'completed'
id: getbranches
uses: asterisk/asterisk-ci-actions/GetCherryPickBranchesFromPR@main
with:
repo: ${{env.REPO}}
pr_number: ${{steps.getpr.outputs.PR_NUMBER}}
cherry_pick_regex: ${{vars.CHERRY_PICK_REGEX}}
github_token: ${{secrets.GITHUB_TOKEN}}
- name: Add cherry-pick reminder
if: github.event.action == 'completed'
uses: actions/github-script@v7
env:
PR_NUMBER: ${{steps.getpr.outputs.PR_NUMBER}}
CHERRY_PICK_REMINDER: ${{vars.CHERRY_PICK_REMINDER}}
BRANCHES_OUTPUT: ${{toJSON(steps.getbranches.outputs)}}
BRANCH_COUNT: ${{steps.getbranches.outputs.branch_count}}
FORCED_NONE: ${{steps.getbranches.outputs.forced_none}}
with:
retries: 5
script: |
if (process.env.FORCED_NONE === 'true' ||
process.env.BRANCH_COUNT > 0) {
core.info("No cherry-pick reminder needed.");
return;
}
let comments = await github.rest.issues.listComments({
issue_number: process.env.PR_NUMBER,
owner: context.repo.owner,
repo: context.repo.repo,
});
let found = false;
for (const c of comments.data) {
if (c.body.startsWith("<!--CPR-->")) {
found = true;
break;
}
}
if (found) {
core.info("Cherry-pick reminder already exists.");
return;
}
core.info("Adding cherry-pick reminder.");
await github.rest.issues.createComment({
issue_number: process.env.PR_NUMBER,
owner: context.repo.owner,
repo: context.repo.repo,
body: process.env.CHERRY_PICK_REMINDER
})
return;
- name: Add reviewers
if: github.event.action == 'completed'
uses: actions/github-script@v7
env:
PR_NUMBER: ${{steps.getpr.outputs.PR_NUMBER}}
REVIEWERS: ${{vars.PR_REVIEWERS}}
with:
retries: 5
script: |
let rs = JSON.parse(process.env.REVIEWERS);
let users = [];
let teams = [];
for (const r of rs) {
if (r.indexOf("/") > 0) {
teams.push(r.split('/')[1]);
} else {
users.push(r);
}
}
if (teams.length > 0 || users.length > 0) {
core.info(`Adding user reviewers ${users}`);
core.info(`Adding team reviewers ${teams}`);
await github.rest.pulls.requestReviewers({
pull_number: process.env.PR_NUMBER,
owner: context.repo.owner,
repo: context.repo.repo,
reviewers: users,
team_reviewers: teams
});
}
return;

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name: PRSubmitTests
run-name: "PR ${{github.event.number}} ${{github.event.action}} by ${{ github.actor }}"
on:
pull_request:
types: [opened, reopened, synchronize]
concurrency:
group: ${{github.workflow}}-${{github.event.number}}
cancel-in-progress: true
env:
ASTERISK_REPO: ${{github.repository}}
PR_NUMBER: ${{github.event.number}}
PR_COMMIT: ${{github.event.pull_request.head.sha}}
BRANCH: ${{github.event.pull_request.base.ref}}
jobs:
#
# Pull requests created from forked respositories don't have access to
# the "Action Variables" ('vars' context) so we need to retrieve control
# data from an action.
#
PRSGetControlData:
runs-on: ubuntu-latest
outputs:
control_data: ${{ steps.setvars.outputs.control_data }}
steps:
- id: setvars
uses: asterisk/asterisk-ci-actions/GetRepoControlData@main
with:
repo: ${{ github.event.repository.name}}
- name: DumpEnvironment
uses: asterisk/asterisk-ci-actions/DumpEnvironmentAction@main
with:
action-inputs: ${{toJSON(inputs)}}
action-vars: ${{ toJSON(steps.setvars.outputs) }}
PRSUnitTests:
needs: PRSGetControlData
runs-on: ubuntu-latest
env:
UNITTEST_COMMAND: ${{ fromJSON(needs.PRSGetControlData.outputs.control_data).UNITTEST_COMMAND }}
steps:
- name: Run Unit Tests
uses: asterisk/asterisk-ci-actions/AsteriskUnitComposite@main
with:
asterisk_repo: ${{env.ASTERISK_REPO}}
pr_number: ${{env.PR_NUMBER}}
base_branch: ${{env.BRANCH}}
unittest_command: ${{env.UNITTEST_COMMAND}}
PRSGateTestMatrix:
runs-on: ubuntu-latest
needs: PRSGetControlData
continue-on-error: false
strategy:
fail-fast: false
matrix:
group: ${{ fromJSON(fromJSON(needs.PRSGetControlData.outputs.control_data).GATETEST_LIST) }}
env:
TESTSUITE_REPO: "${{ fromJSON(needs.PRSGetControlData.outputs.control_data).TESTSUITE_REPO }}"
GATETEST_COMMANDS: "${{ fromJSON(needs.PRSGetControlData.outputs.control_data).GATETEST_COMMANDS }}"
GATETEST_COMMAND: "${{ toJSON(fromJSON(fromJSON(needs.PRSGetControlData.outputs.control_data).GATETEST_COMMANDS)[matrix.group]) }}"
steps:
- id: runtest
name: Run Gate Tests for ${{ matrix.group }}
uses: asterisk/asterisk-ci-actions/AsteriskGateComposite@main
with:
test_type: Gate
asterisk_repo: ${{env.ASTERISK_REPO}}
pr_number: ${{env.PR_NUMBER}}
base_branch: ${{env.BRANCH}}
testsuite_repo: ${{env.TESTSUITE_REPO}}
gatetest_group: ${{matrix.group}}
gatetest_command: ${{env.GATETEST_COMMAND}}
PRSTestResults:
if: always()
runs-on: ubuntu-latest
needs: [PRSUnitTests,PRSGateTestMatrix]
steps:
- name: Check test matrix status
env:
RESULT_UNIT: ${{ needs.PRSUnitTests.result }}
RESULT_GATE: ${{ needs.PRSGateTestMatrix.result }}
run: |
declare -i rc=0
echo "all results: ${{ toJSON(needs.*.result) }}"
case $RESULT_UNIT in
success)
echo "::notice::Unit tests passed"
;;
skipped)
echo "::error::Unit tests were skipped because of an earlier failure"
rc+=1
;;
*)
echo "::error::One or more unit tests failed ($RESULT_UNIT)"
rc+=1
esac
case $RESULT_GATE in
success)
echo "::notice::Gate tests passed"
;;
skipped)
echo "::error::Gate tests were skipped because of an earlier failure"
rc+=1
;;
*)
echo "::error::One or more gate tests failed ($RESULT_GATE)"
rc+=1
esac
echo "::notice::Final result code: $rc"
exit $rc

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name: Releaser
run-name: ${{ github.actor }} is creating ${{vars.PRODUCT_NAME}} release ${{inputs.new_version}}
on:
workflow_dispatch:
inputs:
new_version:
description: |
New Version:
Examples:
20.4.0-rc1, 20.4.0-rc2, 20.4.0, 20.4.1
certified-20.4-cert1-rc1, certified-20.4-cert1
required: true
type: string
is_security:
description: |
Security?
(No prev RCs)
required: true
type: boolean
default: false
advisories:
description: |
Comma separated list of advisories.
NO SPACES
Example: GHSA-4xjp-22g4-9fxm,GHSA-4xjp-22g4-zzzz
required: false
type: string
is_hotfix:
description: |
Hotfix?
(A patch release but not security. No prev RCs)
required: true
type: boolean
default: false
force_cherry_pick:
description: |
Force cherry-pick for non-RC1 releases? USE WITH CAUTION!
required: true
type: boolean
default: false
push_release_branches:
description: |
Push release branches live?
required: true
type: boolean
default: false
create_github_release:
description: |
Create the GitHub release?
required: true
type: boolean
default: false
push_tarballs:
description: |
Push tarballs to downloads server?
required: true
type: boolean
default: false
send_email:
description: |
Send announcement emails?
required: true
type: boolean
default: false
jobs:
ReleaseAsterisk:
runs-on: ubuntu-latest
steps:
- name: Run Releaser
uses: asterisk/asterisk-ci-actions/ReleaserComposite@main
with:
product: ${{vars.PRODUCT_NAME}}
is_security: ${{inputs.is_security}}
advisories: ${{inputs.advisories}}
is_hotfix: ${{inputs.is_hotfix}}
new_version: ${{inputs.new_version}}
force_cherry_pick: ${{inputs.force_cherry_pick}}
push_release_branches: ${{inputs.push_release_branches}}
create_github_release: ${{inputs.create_github_release}}
push_tarballs: ${{inputs.push_tarballs}}
send_email: ${{inputs.send_email}}
repo: ${{github.repository}}
mail_list_ga: ${{vars.MAIL_LIST_GA}}
mail_list_rc: ${{vars.MAIL_LIST_RC}}
mail_list_cert_ga: ${{vars.MAIL_LIST_CERT_GA}}
mail_list_cert_rc: ${{vars.MAIL_LIST_CERT_RC}}
mail_list_sec: ${{vars.MAIL_LIST_SEC_ADV}}
sec_adv_url_base: ${{vars.SEC_ADV_URL_BASE}}
gpg_private_key: ${{secrets.ASTDEV_GPG_PRIV_KEY}}
github_token: ${{secrets.GITHUB_TOKEN}}
application_id: ${{secrets.ASTERISK_ORG_ACCESS_APP_ID}}
application_private_key: ${{secrets.ASTERISK_ORG_ACCESS_APP_PRIV_KEY}}
asteriskteamsa_username: ${{secrets.ASTERISKTEAMSA_GMAIL_ACCT}}
asteriskteamsa_token: ${{secrets.ASTERISKTEAMSA_GMAIL_TOKEN}}
deploy_ssh_priv_key: ${{secrets.DOWNLOADS_DEPLOY_SSH_PRIV_KEY}}
deploy_ssh_username: ${{secrets.DOWNLOADS_DEPLOY_SSH_USERNAME}}
deploy_host: ${{vars.DEPLOY_HOST}}
deploy_dir: ${{vars.DEPLOY_DIR}}

1
.gitignore vendored
View File

@ -38,3 +38,4 @@ out/
*.orig
tests/CI/output
.develvars
configure

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@ -1,11 +0,0 @@
[gerrit]
defaultbranch=master
basebranch=master
#
# Intentional padding to ensure it is possible to point a commit
# to an alternative gerrit server/repository without breaking
# cherry-pick between branches.
#
host=gerrit.asterisk.org
port=29418
project=asterisk.git

1
.lastclean Normal file
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@ -0,0 +1 @@
40

1
.version Normal file
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@ -0,0 +1 @@
20.7.0

10
BUGS
View File

@ -1,22 +1,22 @@
Asterisk Bug Tracking Information
=================================
To learn about and report Asterisk bugs, please visit
To learn about and report Asterisk bugs, please visit
the official Asterisk Bug Tracker at:
https://issues.asterisk.org/jira
https://github.com/asterisk/asterisk/issues/
For more information on using the bug tracker, or to
For more information on using the bug tracker, or to
learn how you can contribute by acting as a bug marshal
please see:
http://www.asterisk.org/developers/bug-guidelines
https://docs.asterisk.org/Asterisk-Community/Asterisk-Issue-Guidelines/
If you would like to submit a feature request, please
resist the temptation to post it to the bug tracker.
Feature requests should be posted to the asterisk-dev
mailing list, located at:
http://lists.digium.com
http://lists.digium.com
Thank you!

1
CHANGES.md Symbolic link
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@ -0,0 +1 @@
ChangeLogs/ChangeLog-20.7.0.md

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@ -0,0 +1,741 @@
Change Log for Release 20.3.0
========================================
Summary:
----------------------------------------
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
User Notes:
----------------------------------------
- ### cel: add local optimization begin event
The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
- ### chan_dahdi: Add dialmode option for FXS lines.
A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.
Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.
- ### pbx_dundi: Add PJSIP support.
DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.
- ### cli: increase channel column width
This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
- ### app_senddtmf: Add SendFlash AMI action.
The SendFlash AMI action now allows sending
a hook flash event on a channel.
- ### res_http_media_cache: Introduce options and customize
The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
* timeout_secs
* user_agent
* follow_location
* max_redirects
* protocols
* redirect_protocols
* dns_cache_timeout_secs
- ### test.c: Fix counting of tests and add 2 new tests
The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.
- ### bridge_builtin_features: add beep via touch variable
Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.
- ### format_sln: add .slin as supported file extension
format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.
Upgrade Notes:
----------------------------------------
- ### cel: add local optimization begin event
The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.
Closed Issues:
----------------------------------------
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
Commits By Author:
----------------------------------------
- ### Asterisk Development Team (1):
- Update for 20.3.0-rc1
- ### Fabrice Fontaine (2):
- main/iostream.c: fix build with libressl
- configure: fix detection of re-entrant resolver functions
- ### George Joseph (13):
- make_version: Strip svn stuff and suppress ref HEAD errors
- test.c: Fix counting of tests and add 2 new tests
- Initial GitHub Issue Templates
- Initial GitHub PRs
- .github: Remove unnecessary parameter in CherryPickTest
- .github: Update issue templates
- .github: Add cherry-pick test progress labels
- .github: Refactor CP progress and add new PR test progress
- .github: Remove separate set labels step from new PR
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Fix CherryPickTest to only run when it should
- .github: Add AsteriskReleaser
- Set up new ChangeLogs directory
- ### Henning Westerholt (2):
- chan_pjsip: fix music on hold continues after INVITE with replaces
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- ### Holger Hans Peter Freyther (1):
- res_http_media_cache: Introduce options and customize
- ### Jaco Kroon (2):
- app_queue: periodic announcement configurable start time.
- res_calendar: output busy state as part of show calendar.
- ### Joshua C. Colp (2):
- pbx_dundi: Fix PJSIP endpoint configuration check.
- LICENSE: Update link to trademark policy.
- ### Joshua Colp (1):
- Revert "app_queue: periodic announcement configurable start time."
- ### Maximilian Fridrich (1):
- res_pjsip: mediasec: Add Security-Client headers after 401
- ### Mike Bradeen (5):
- cli: increase channel column width
- format_sln: add .slin as supported file extension
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- bridge_builtin_features: add beep via touch variable
- cel: add local optimization begin event
- ### Naveen Albert (8):
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- app_senddtmf: Add SendFlash AMI action.
- func_json: Fix JSON parsing issues.
- app_queue: Fix minor xmldoc duplication and vagueness.
- voicemail.conf: Fix incorrect comment about #include.
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- chan_dahdi: Add dialmode option for FXS lines.
- ### Sean Bright (5):
- contrib: rc.archlinux.asterisk uses invalid redirect.
- res_agi: RECORD FILE plays 2 beeps.
- ael: Regenerate lexers and parsers.
- loader.c: Minor module key check simplification.
- core: Cleanup gerrit and JIRA references. (#57)
- ### The_Blode (1):
- install_prereq: Add Linux Mint support.
Detail:
----------------------------------------
- ### Set up new ChangeLogs directory
Author: George Joseph
Date: 2023-05-09
- ### .github: Add AsteriskReleaser
Author: George Joseph
Date: 2023-05-05
- ### chan_pjsip: also return all codecs on empty re-INVITE for late offers
Author: Henning Westerholt
Date: 2023-05-03
We should also return all codecs on an re-INVITE without SDP for a
call that used late offer (e.g. no SDP in the initial INVITE, SDP
in the ACK). Bugfix for feature introduced in ASTERISK-30193
(https://issues.asterisk.org/jira/browse/ASTERISK-30193)
Migration from previous gerrit change that was not merged.
- ### cel: add local optimization begin event
Author: Mike Bradeen
Date: 2023-05-02
The current AST_CEL_LOCAL_OPTIMIZE event is and has been
triggered on a local optimization end to serve as a flag
indicating the event occurred. This change adds a second
AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail.
Resolves: #52
UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.
UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
- ### core: Cleanup gerrit and JIRA references. (#57)
Author: Sean Bright
Date: 2023-05-03
* 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
- ### .github: Fix CherryPickTest to only run when it should
Author: George Joseph
Date: 2023-05-03
Fixed CherryPickTest so it triggers only on the
"cherry-pick-test" label instead of all labels.
- ### .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
Author: George Joseph
Date: 2023-05-02
- ### .github: Remove separate set labels step from new PR
Author: George Joseph
Date: 2023-05-02
- ### .github: Refactor CP progress and add new PR test progress
Author: George Joseph
Date: 2023-05-02
- ### res_pjsip: mediasec: Add Security-Client headers after 401
Author: Maximilian Fridrich
Date: 2023-05-02
When using mediasec, requests sent after a 401 must still contain the
Security-Client header according to
draft-dawes-sipcore-mediasec-parameter.
Resolves: #48
- ### .github: Add cherry-pick test progress labels
Author: George Joseph
Date: 2023-05-02
- ### LICENSE: Update link to trademark policy.
Author: Joshua C. Colp
Date: 2023-05-01
Resolves: #43
- ### chan_dahdi: Add dialmode option for FXS lines.
Author: Naveen Albert
Date: 2023-04-28
Currently, both pulse and tone dialing are always enabled
on all FXS lines, with no way of disabling one or the other.
In some circumstances, it is desirable or necessary to
disable one of these, and this behavior can be problematic.
A new "dialmode" option is added which allows setting the
methods to support on a per channel basis for FXS (FXO
signalled lines). The four options are "both", "pulse",
"dtmf"/"tone", and "none".
Additionally, integration with the CHANNEL function is
added so that this setting can be updated for a channel
during a call.
Resolves: #35
ASTERISK-29992
UserNote: A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.
Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.
- ### .github: Update issue templates
Author: George Joseph
Date: 2023-05-01
- ### .github: Remove unnecessary parameter in CherryPickTest
Author: George Joseph
Date: 2023-05-01
- ### Initial GitHub PRs
Author: George Joseph
Date: 2023-04-28
- ### Initial GitHub Issue Templates
Author: George Joseph
Date: 2023-04-28
- ### pbx_dundi: Fix PJSIP endpoint configuration check.
Author: Joshua C. Colp
Date: 2023-04-13
ASTERISK-28233
- ### Revert "app_queue: periodic announcement configurable start time."
Author: Joshua Colp
Date: 2023-04-11
This reverts commit 3fd0b65bae4b1b14434737ffcf0da4aa9ff717f6.
Reason for revert: Causes segmentation fault.
- ### res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
Author: Naveen Albert
Date: 2023-02-17
The current STIR/SHAKEN signing process is inconsistent with the
RFCs in a couple ways that can cause interoperability issues.
RFC8225 specifies that the keys must be ordered lexicographically, but
currently the fields are simply ordered according to the order
in which they were added to the JSON object, which is not
compliant with the RFC and can cause issues with some carriers.
To fix this, we now leverage libjansson's ability to dump a JSON
object sorted by key value, yielding the correct field ordering.
Additionally, telephone numbers must have any leading + prefix removed
and must not contain characters outside of 0-9, *, and # in order
to comply with the RFCs. Numbers are now properly formatted as such.
ASTERISK-30407 #close
- ### pbx_dundi: Add PJSIP support.
Author: Naveen Albert
Date: 2022-12-09
Adds PJSIP as a supported technology to DUNDi.
To facilitate this, we now allow an endpoint to be specified
for outgoing PJSIP calls. We also allow users to force a specific
channel technology for outgoing SIP-protocol calls.
ASTERISK-28109 #close
ASTERISK-28233 #close
- ### install_prereq: Add Linux Mint support.
Author: The_Blode
Date: 2023-03-17
ASTERISK-30359 #close
- ### chan_pjsip: fix music on hold continues after INVITE with replaces
Author: Henning Westerholt
Date: 2023-03-21
In a three party scenario with INVITE with replaces, we need to
unhold the call, otherwise one party continues to get music on
hold, and the call is not properly bridged between them.
ASTERISK-30428
- ### voicemail.conf: Fix incorrect comment about #include.
Author: Naveen Albert
Date: 2023-03-28
A comment at the top of voicemail.conf says that #include
cannot be used in voicemail.conf because this breaks
the ability for app_voicemail to auto-update passwords.
This is factually incorrect, since Asterisk has no problem
updating files that are #include'd in the main configuration
file, and this does work in voicemail.conf as well.
ASTERISK-30479 #close
- ### app_queue: Fix minor xmldoc duplication and vagueness.
Author: Naveen Albert
Date: 2023-04-03
The F option in the xmldocs for the Queue application
was erroneously duplicated, causing it to display
twice on the wiki. The two sections are now merged into one.
Additionally, the description for the d option was quite
vague. Some more details are added to provide context
as to what this actually does.
ASTERISK-30486 #close
- ### test.c: Fix counting of tests and add 2 new tests
Author: George Joseph
Date: 2023-03-28
The unit test XML output was counting all registered tests as "run"
even when only a subset were actually requested to be run and
the "failures" attribute was missing.
* The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
* The "failures" attribute was added to the "testsuite"
element.
Also added 2 new unit tests that just pass and fail to be
used for CI testing.
- ### res_calendar: output busy state as part of show calendar.
Author: Jaco Kroon
Date: 2023-03-23
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
- ### loader.c: Minor module key check simplification.
Author: Sean Bright
Date: 2023-03-23
- ### ael: Regenerate lexers and parsers.
Author: Sean Bright
Date: 2023-03-21
Various changes to ensure that the lexers and parsers can be correctly
generated when REBUILD_PARSERS is enabled.
Some notes:
* Because of the version of flex we are using to generate the lexers
(2.5.35) some post-processing in the Makefile is still required.
* The generated lexers do not contain the problematic C99 check that
was being replaced by the call to sed in the respective Makefiles so
it was removed.
* Since these files are generated, they will include trailing
whitespace in some places. This does not need to be corrected.
- ### bridge_builtin_features: add beep via touch variable
Author: Mike Bradeen
Date: 2023-03-01
Add periodic beep option to one-touch recording by setting
the touch variable TOUCH_MONITOR_BEEP or
TOUCH_MIXMONITOR_BEEP to the desired interval in seconds.
If the interval is less than 5 seconds, a minimum of 5
seconds will be imposed. If the interval is set to an
invalid value, it will default to 15 seconds.
A new test event PERIODIC_HOOK_ENABLED was added to the
func_periodic_hook hook_on function to indicate when
a hook is started. This is so we can test that the touch
variable starts the hook as expected.
ASTERISK-30446
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
Author: Mike Bradeen
Date: 2023-03-13
While it is possible to create multiple mixmonitor instances
on a channel, it was not previously possible to mute individual
instances.
This change includes the ability to specify the MixMonitorID
when calling the manager action: MixMonitorMute. This will
allow an individual MixMonitor instance to be muted via id.
This id can be stored as a channel variable using the 'i'
MixMonitor option.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor spy-type audiohooks on the channel.
This is done via the new audiohook function:
ast_audiohook_set_mute_all.
ASTERISK-30464
- ### format_sln: add .slin as supported file extension
Author: Mike Bradeen
Date: 2023-03-14
Adds '.slin' to existing supported file extensions:
.sln and .raw
ASTERISK-30465
- ### res_agi: RECORD FILE plays 2 beeps.
Author: Sean Bright
Date: 2023-03-08
Sending the "RECORD FILE" command without the optional
`offset_samples` argument can result in two beeps playing on the
channel.
This bug has been present since Asterisk 0.3.0 (2003-02-06).
ASTERISK-30457 #close
- ### func_json: Fix JSON parsing issues.
Author: Naveen Albert
Date: 2023-02-26
Fix issue with returning empty instead of dumping
the JSON string when recursing.
Also adds a unit test to capture this fix.
ASTERISK-30441 #close
- ### app_senddtmf: Add SendFlash AMI action.
Author: Naveen Albert
Date: 2023-02-26
Adds an AMI action to send a flash event
on a channel.
ASTERISK-30440 #close
- ### app_dial: Fix DTMF not relayed to caller on unanswered calls.
Author: Naveen Albert
Date: 2023-03-04
DTMF frames are not handled in app_dial when sent towards the
caller. This means that if DTMF is sent to the calling party
and the call has not yet been answered, the DTMF is not audible.
This is now fixed by relaying DTMF frames if only a single
destination is being dialed.
ASTERISK-29516 #close
- ### configure: fix detection of re-entrant resolver functions
Author: Fabrice Fontaine
Date: 2023-03-08
uClibc does not provide res_nsearch:
asterisk-16.0.0/main/dns.c:506: undefined reference to `res_nsearch'
Patch coded by Yann E. MORIN:
http://lists.busybox.net/pipermail/buildroot/2018-October/232630.html
ASTERISK-21795 #close
Signed-off-by: Bernd Kuhls <bernd.kuhls@t-online.de>
[Retrieved from:
https: //git.buildroot.net/buildroot/tree/package/asterisk/0005-configure-fix-detection-of-re-entrant-resolver-funct.patch]
Signed-off-by: Fabrice Fontaine <fontaine.fabrice@gmail.com>
- ### cli: increase channel column width
Author: Mike Bradeen
Date: 2023-03-06
For 'core show channels', the Channel name field is increased
to 64 characters and the Location name field is increased to
32 characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
ASTERISK-30455
- ### app_queue: periodic announcement configurable start time.
Author: Jaco Kroon
Date: 2023-02-21
This newly introduced periodic-announce-startdelay makes it possible to
configure the initial start delay of the first periodic announcement
after which periodic-announce-frequency takes over.
ASTERISK-30437 #close
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
- ### make_version: Strip svn stuff and suppress ref HEAD errors
Author: George Joseph
Date: 2023-03-13
* All of the code that used subversion has been removed.
* When Asterisk is checked out from a tag or commit instead
of one of the regular branches, git would emit messages like
"fatal: ref HEAD is not a symbolic ref" which weren't fatal
at all. Those are now suppressed.
- ### res_http_media_cache: Introduce options and customize
Author: Holger Hans Peter Freyther
Date: 2022-10-16
Make the existing CURL parameters configurable and allow
to specify the usable protocols, proxy and DNS timeout.
ASTERISK-30340
- ### main/iostream.c: fix build with libressl
Author: Fabrice Fontaine
Date: 2023-02-25
Fix the following build failure with libressl by using SSL_is_server
which is available since version 2.7.0 and
https://github.com/libressl-portable/openbsd/commit/d7ec516916c5eaac29b02d7a8ac6570f63b458f7:
iostream.c: In function 'ast_iostream_close':
iostream.c:559:41: error: invalid use of incomplete typedef 'SSL' {aka 'struct ssl_st'}
559 | if (!stream->ssl->server) {
| ^~
ASTERISK-30107 #close
Fixes: - http://autobuild.buildroot.org/results/ce4d62d00bb77ba5b303cacf6be7e350581a62f9
- ### contrib: rc.archlinux.asterisk uses invalid redirect.
Author: Sean Bright
Date: 2023-03-02
`rc.archlinux.asterisk`, which explicitly requests bash in its
shebang, uses the following command syntax:
${DAEMON} -rx "core stop now" > /dev/null 2&>1
The intent of which is to execute:
${DAEMON} -rx "core stop now"
While sending both stdout and stderr to `/dev/null`. Unfortunately,
because the `&` is in the wrong place, bash is interpreting the `2` as
just an additional argument to the `$DAEMON` command and not as a file
descriptor and proceeds to use the bashism `&>` to send stderr and
stdout to a file named `1`.
So we clean it up and just use bash's shortcut syntax.
Issue raised and a fix suggested (but not used) by peutch on GitHub¹.
ASTERISK-30449 #close
1. https://github.com/asterisk/asterisk/pull/31

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Change Log for Release 20.3.1
========================================
Links:
----------------------------------------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
Summary:
----------------------------------------
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
User Notes:
----------------------------------------
- ### res_http_media_cache: Introduce options and customize
The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
* timeout_secs
* user_agent
* follow_location
* max_redirects
* protocols
* redirect_protocols
* dns_cache_timeout_secs
- ### format_sln: add .slin as supported file extension
format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.
- ### bridge_builtin_features: add beep via touch variable
Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.
- ### app_senddtmf: Add SendFlash AMI action.
The SendFlash AMI action now allows sending
a hook flash event on a channel.
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.
- ### pbx_dundi: Add PJSIP support.
DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.
- ### test.c: Fix counting of tests and add 2 new tests
The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.
- ### cli: increase channel column width
This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
Upgrade Notes:
----------------------------------------
Closed Issues:
----------------------------------------
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying
Commits By Author:
----------------------------------------
- ### George Joseph (1):
- apply_patches: Sort patch list before applying
- ### Sean Bright (1):
- apply_patches: Use globbing instead of file/sort.
- ### Stanislav Abramenkov (1):
- pjsip: Upgrade bundled version to pjproject 2.13.1
Detail:
----------------------------------------
- ### apply_patches: Use globbing instead of file/sort.
Author: Sean Bright
Date: 2023-07-06
This accomplishes the same thing as a `find ... | sort` but with the
added benefit of clarity and avoiding a call to a subshell.
Additionally drop the -s option from call to patch as it is not POSIX.
- ### apply_patches: Sort patch list before applying
Author: George Joseph
Date: 2023-07-06
The apply_patches script wasn't sorting the list of patches in
the "patches" directory before applying them. This left the list
in an indeterminate order. In most cases, the list is actually
sorted but rarely, they can be out of order and cause dependent
patches to fail to apply.
We now sort the list but the "sort" program wasn't in the
configure scripts so we needed to add that and regenerate
the scripts as well.
Resolves: #193
- ### pjsip: Upgrade bundled version to pjproject 2.13.1
Author: Stanislav Abramenkov
Date: 2023-07-05

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Change Log for Release 20.4.0
========================================
Links:
----------------------------------------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
Summary:
----------------------------------------
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
User Notes:
----------------------------------------
- ### AMI: Add parking position parameter to Park action
New ParkingSpace parameter has been added to AMI action Park.
- ### res_musiconhold: Add option to loop last file.
The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.
- ### AMI: Add CoreShowChannelMap action.
New AMI action CoreShowChannelMap has been added.
- ### sig_analog: Add fuller Caller ID support.
Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
When creating a bridge using the ARI the 'type' argument now
accepts a new value 'sdp_label' which will configure the bridge to add
labels for each stream in the SDP with the corresponding channel id.
- ### app_queue: Preserve reason for realtime queues
Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.
Upgrade Notes:
----------------------------------------
- ### app_queue: Preserve reason for realtime queues
Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.
Closed Issues:
----------------------------------------
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.
Commits By Author:
----------------------------------------
- ### Asterisk Development Team (2):
- Update for 20.4.0-rc1
- Update for 20.4.0-rc2
- ### Ben Ford (2):
- AMI: Add CoreShowChannelMap action.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- ### George Joseph (23):
- .github: Add cherry-pick reminder to new PRs
- .github: Fix quoting in PROpenedOrUpdated
- .github: Don't add cherry-pick reminder if it's already present
- .github: Change title of AsteriskReleaser job
- .github: Update field descriptions for AsteriskReleaser
- .github Ignore error when adding reviewrs to PR
- .github: Fix issues with cherry-pick-reminder
- .github: Rework for merge approval
- build: Fix a few gcc 13 issues
- test_stasis_endpoints.c: Make channel_messages more stable
- test_statis_endpoints: Fix channel_messages test again
- .github: Allow PR submit checks to be re-run by label
- .github: Fix recheck label manipulation
- .github: Fix recheck label typos
- .github: Add recheck label test to additional jobs
- .github: Fix use of 'contains'
- .github: Remove result check from PROpenUpdateGateTests
- rest-api: Updates for new documentation site
- .github: Move publish docs to new file CreateDocs.yml
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Updates for AsteriskReleaser
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- app.h: Move declaration of ast_getdata_result before its first use
- ### Gitea (1):
- .github: Tweak new feature language, and move feature requests elsewhere.
- ### Jaco Kroon (2):
- configure: fix test code to match gethostbyname_r prototype.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- ### Jiajian Zhou (1):
- AMI: Add parking position parameter to Park action
- ### Joe Searle (1):
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- ### Joshua C. Colp (2):
- .github: Fix staleness check to only run on certain labels.
- .github: Tweak improvement issue type language.
- ### Maximilian Fridrich (1):
- chan_pjsip: Allow topology/session refreshes in early media state
- ### Miguel Angel Nubla (1):
- configure: Makefile downloader enable follow redirects.
- ### Mike Bradeen (4):
- indications: logging changes
- utils: add lock timestamps for DEBUG_THREADS
- res_musiconhold: avoid moh state access on unlocked chan
- app_voicemail: fix imap compilation errors
- ### Nathan Bruning (1):
- app_queue: Add force_longest_waiting_caller option.
- ### Naveen Albert (7):
- asterisk.c: Fix option warning for remote console.
- chan_dahdi: Fix broken hidecallerid setting.
- logrotate: Fix duplicate log entries.
- callerid: Allow specifying timezone for date/time.
- sig_analog: Add fuller Caller ID support.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- res_musiconhold: Add option to loop last file.
- ### Niklas Larsson (1):
- app_queue: Preserve reason for realtime queues
- ### Olaf Titz (1):
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- ### Sean Bright (10):
- xml.c: Process XML Inclusions recursively.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- sounds: Update download URL to use HTTPS.
- ast-db-manage: Fix alembic branching error caused by #122.
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- ast-db-manage: Synchronize revisions between comments and code.
- configure: Remove obsolete and deprecated constructs.
- res_crypto.c: Gracefully handle potential key filename truncation.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- ### alex2grad (1):
- app_followme: fix issue with enable_callee_prompt=no (#88)
- ### zhengsh (1):
- res_sorcery_memory_cache.c: Fix memory leak
- ### zhou_jiajian (1):
- res_fax_spandsp.c: Clean up a spaces/tabs issue
Detail:
----------------------------------------
- ### app.h: Move declaration of ast_getdata_result before its first use
Author: George Joseph
Date: 2023-07-10
The ast_app_getdata() and ast_app_getdata_terminator() declarations
in app.h were changed recently to return enum ast_getdata_result
(which is how they were defined in app.c). The existing
declaration of ast_getdata_result in app.h was about 1000 lines
after those functions however so under certain circumstances,
a "use before declaration" error was thrown by the compiler.
The declaration of the enum was therefore moved to before those
functions.
Resolves: #200
- ### doc: Remove obsolete CHANGES-staging and UPGRADE-staging
Author: George Joseph
Date: 2023-07-10
- ### .github: Updates for AsteriskReleaser
Author: George Joseph
Date: 2023-06-30
- ### app_voicemail: fix imap compilation errors
Author: Mike Bradeen
Date: 2023-06-26
Fixes two compilation errors in app_voicemail_imap, one due to an unsed
variable and one due to a new variable added in the incorrect location
in _163.
Resolves: #174
- ### res_musiconhold: avoid moh state access on unlocked chan
Author: Mike Bradeen
Date: 2023-05-31
Move channel unlock to after moh state access to avoid
potential unlocked access to state.
Resolves: #133
- ### utils: add lock timestamps for DEBUG_THREADS
Author: Mike Bradeen
Date: 2023-05-23
Adds last locked and unlocked timestamps as well as a
counter for the number of times the lock has been
attempted (vs locked/unlocked) to debug output printed
using the DEBUG_THREADS option.
Resolves: #110
- ### .github: Back out triggering PROpenedOrUpdated by label
Author: George Joseph
Date: 2023-06-29
- ### .github: Move publish docs to new file CreateDocs.yml
Author: George Joseph
Date: 2023-06-27
- ### rest-api: Updates for new documentation site
Author: George Joseph
Date: 2023-06-26
The new documentation site uses traditional markdown instead
of the Confluence flavored version. This required changes in
the mustache templates and the python that generates the files.
- ### .github: Remove result check from PROpenUpdateGateTests
Author: George Joseph
Date: 2023-06-27
- ### .github: Fix use of 'contains'
Author: George Joseph
Date: 2023-06-26
- ### .github: Add recheck label test to additional jobs
Author: George Joseph
Date: 2023-06-26
- ### .github: Fix recheck label typos
Author: George Joseph
Date: 2023-06-26
- ### .github: Fix recheck label manipulation
Author: George Joseph
Date: 2023-06-26
- ### .github: Allow PR submit checks to be re-run by label
Author: George Joseph
Date: 2023-06-26
- ### app_voicemail_imap: Fix message count when IMAP server is unavailable
Author: Olaf Titz
Date: 2023-06-15
Some callers of __messagecount did not correctly handle error return,
instead returning a -1 message count.
This caused a notification with "Messages-Waiting: yes" and
"Voice-Message: -1/0 (0/0)" if the IMAP server was unavailable.
Fixes: #64
- ### res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
Author: Sean Bright
Date: 2023-06-12
Resolves: #116
- ### res_pjsip_session: Added new function calls to avoid ABI issues.
Author: Ben Ford
Date: 2023-06-05
Added two new functions (ast_sip_session_get_dialog and
ast_sip_session_get_pjsip_inv_state) that retrieve the dialog and the
pjsip_inv_state respectively from the pjsip_inv_session on the
ast_sip_session struct. This is due to pjproject adding a new field to
the pjsip_inv_session struct that caused crashes when trying to access
fields that were no longer where they were expected to be if a module
was compiled against a different version of pjproject.
Resolves: #145
- ### app_queue: Add force_longest_waiting_caller option.
Author: Nathan Bruning
Date: 2023-01-24
This adds an option 'force_longest_waiting_caller' which changes the
global behavior of the queue engine to prevent queue callers from
'jumping ahead' when an agent is in multiple queues.
Resolves: #108
Also closes old asterisk issues:
- ASTERISK-17732
- ASTERISK-17570
- ### pjsip_transport_events.c: Use %zu printf specifier for size_t.
Author: Sean Bright
Date: 2023-06-05
Partially resolves #143.
- ### res_crypto.c: Gracefully handle potential key filename truncation.
Author: Sean Bright
Date: 2023-06-05
Partially resolves #143.
- ### configure: Remove obsolete and deprecated constructs.
Author: Sean Bright
Date: 2023-06-01
These were uncovered when trying to run `bootstrap.sh` with Autoconf
2.71:
* AC_CONFIG_HEADER() is deprecated in favor of AC_CONFIG_HEADERS().
* AC_HEADER_TIME is obsolete.
* $as_echo is deprecated in favor of AS_ECHO() which requires an update
to ax_pthread.m4.
Note that the generated artifacts in this commit are from Autoconf 2.69.
Resolves #139
- ### res_fax_spandsp.c: Clean up a spaces/tabs issue
Author: zhou_jiajian
Date: 2023-05-26
- ### ast-db-manage: Synchronize revisions between comments and code.
Author: Sean Bright
Date: 2023-06-06
In a handful of migrations, the comment header that indicates the
current and previous revisions has drifted from the identifiers
revision and down_revision variables. This updates the comment headers
to match the code.
- ### test_statis_endpoints: Fix channel_messages test again
Author: George Joseph
Date: 2023-06-12
- ### res_crypto.c: Avoid using the non-portable ALLPERMS macro.
Author: Sean Bright
Date: 2023-06-05
ALLPERMS is not POSIX and it's trivial enough to not jump through
autoconf hoops to check for it.
Fixes #149.
- ### tcptls: when disabling a server port, we should set the accept_fd to -1.
Author: Jaco Kroon
Date: 2023-06-02
If we don't set this to -1 if the structure can be potentially re-used
later then it's possible that we'll issue a close() on an unrelated file
descriptor, breaking asterisk in other interesting ways.
I believe this to be an unlikely scenario, but it costs nothing to be
safe.
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
- ### AMI: Add parking position parameter to Park action
Author: Jiajian Zhou
Date: 2023-05-19
Add a parking space extension parameter (ParkingSpace) to the Park action.
Park action will attempt to park the call to that extension.
If the extension is already in use, then execution will continue at the next priority.
UserNote: New ParkingSpace parameter has been added to AMI action Park.
- ### test_stasis_endpoints.c: Make channel_messages more stable
Author: George Joseph
Date: 2023-06-09
The channel_messages test was assuming that stasis would return
messages in a specific order. This is an incorrect assumption as
message ordering was never guaranteed. This was causing the test
to fail occasionally. We now test all the messages for the
required message types instead of testing one by one.
Resolves: #158
- ### build: Fix a few gcc 13 issues
Author: George Joseph
Date: 2023-06-09
* gcc 13 is now catching when a function is declared as returning
an enum but defined as returning an int or vice versa. Fixed
a few in app.h, loader.c, stasis_message.c.
* gcc 13 is also now (incorrectly) complaining of dangling pointers
when assigning a pointer to a local char array to a char *. Had
to change that to an ast_alloca.
Resolves: #155
- ### .github: Rework for merge approval
Author: George Joseph
Date: 2023-06-06
- ### ast-db-manage: Fix alembic branching error caused by #122.
Author: Sean Bright
Date: 2023-06-05
Fixes #147.
- ### app_followme: fix issue with enable_callee_prompt=no (#88)
Author: alex2grad
Date: 2023-06-05
* app_followme: fix issue with enable_callee_prompt=no
If the FollowMe option 'enable_callee_prompt' is set to 'no' then Asterisk
incorrectly sets a winner channel to the channel from which any control frame was read.
This fix sets the winner channel only to the answered channel.
Resolves: #87
ASTERISK-30326
- ### sounds: Update download URL to use HTTPS.
Author: Sean Bright
Date: 2023-06-01
Related to #136
- ### configure: Makefile downloader enable follow redirects.
Author: Miguel Angel Nubla
Date: 2023-06-01
If curl is used for building, any download such as a sounds package
will fail to follow HTTP redirects and will download wrong data.
Resolves: #136
- ### res_musiconhold: Add option to loop last file.
Author: Naveen Albert
Date: 2023-05-25
Adds the loop_last option to res_musiconhold,
which allows the last audio file in the directory
to be looped perpetually once reached, rather than
circling back to the beginning again.
Resolves: #122
ASTERISK-30462
UserNote: The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.
- ### chan_dahdi: Fix Caller ID presentation for FXO ports.
Author: Naveen Albert
Date: 2023-05-25
Currently, the presentation for incoming channels is
always available, because it is never actually set,
meaning the channel presentation can be nonsensical.
If the presentation from the incoming Caller ID spill
is private or unavailable, we now update the channel
presentation to reflect this.
Resolves: #120
ASTERISK-30333
ASTERISK-21741
- ### AMI: Add CoreShowChannelMap action.
Author: Ben Ford
Date: 2023-05-18
Adds a new AMI action (CoreShowChannelMap) that takes in a channel name
and provides a list of all channels that are connected to that channel,
following local channel connections as well.
Resolves: #104
UserNote: New AMI action CoreShowChannelMap has been added.
- ### sig_analog: Add fuller Caller ID support.
Author: Naveen Albert
Date: 2023-05-18
A previous change, ASTERISK_29991, made it possible
to send additional Caller ID parameters that were
not previously supported.
This change adds support for analog DAHDI channels
to now be able to receive these parameters for
on-hook Caller ID, in order to enhance the usability
of CPE that support these parameters.
Resolves: #94
ASTERISK-30331
UserNote: Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
Author: Joe Searle
Date: 2023-05-25
Add new type 'sdp_label' when creating a bridge using the ARI. This will
add labels to the SDP for each stream, the label is set to the
corresponding channel id.
Resolves: #91
UserNote: When creating a bridge using the ARI the 'type' argument now
accepts a new value 'sdp_label' which will configure the bridge to add
labels for each stream in the SDP with the corresponding channel id.
- ### app_queue: Preserve reason for realtime queues
Author: Niklas Larsson
Date: 2023-05-05
When Asterisk is restarted it does not preserve paused reason for
members of realtime queues. This was fixed for non-realtime queues in
ASTERISK_25732
Resolves: #66
UpgradeNote: Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.
UserNote: Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.
- ### .github: Fix issues with cherry-pick-reminder
Author: George Joseph
Date: 2023-06-05
- ### indications: logging changes
Author: Mike Bradeen
Date: 2023-05-16
Increase verbosity to indicate failure due to missing country
and to specify default on CLI dump
Resolves: #89
- ### .github Ignore error when adding reviewrs to PR
Author: George Joseph
Date: 2023-06-05
- ### .github: Update field descriptions for AsteriskReleaser
Author: George Joseph
Date: 2023-05-26
- ### callerid: Allow specifying timezone for date/time.
Author: Naveen Albert
Date: 2023-05-18
The Caller ID generation routine currently is hardcoded
to always use the system time zone. This makes it possible
to optionally specify any TZ-format time zone.
Resolves: #98
ASTERISK-30330
- ### logrotate: Fix duplicate log entries.
Author: Naveen Albert
Date: 2023-05-18
The Asterisk logrotate script contains explicit
references to files with the .log extension,
which are also included when *log is expanded.
This causes issues with newer versions of logrotate.
This fixes this by ensuring that a log file cannot
be referenced multiple times after expansion occurs.
Resolves: #96
ASTERISK-30442
Reported by: EN Barnett
Tested by: EN Barnett
- ### chan_pjsip: Allow topology/session refreshes in early media state
Author: Maximilian Fridrich
Date: 2023-05-10
With this change, session modifications in the early media state are
possible if the SDP was sent reliably and confirmed by a PRACK. For
details, see RFC 6337, escpecially section 3.2.
Resolves: #73
- ### chan_dahdi: Fix broken hidecallerid setting.
Author: Naveen Albert
Date: 2023-05-18
The hidecallerid setting in chan_dahdi.conf currently
is broken for a couple reasons.
First, the actual code in sig_analog to "allow" or "block"
Caller ID depending on this setting improperly used
ast_set_callerid instead of updating the presentation.
This issue was mostly fixed in ASTERISK_29991, and that
fix is carried forward to this code as well.
Secondly, the hidecallerid setting is set on the DAHDI
pvt but not carried forward to the analog pvt properly.
This is because the chan_dahdi config loading code improperly
set permhidecallerid to permhidecallerid from the config file,
even though hidecallerid is what is actually set from the config
file. (This is done correctly for call waiting, a few lines above.)
This is fixed to read the proper value.
Thirdly, in sig_analog, hidecallerid is set to permhidecallerid
only on hangup. This can lead to potential security vulnerabilities
as an allowed Caller ID from an initial call can "leak" into subsequent
calls if no hangup occurs between them. This is fixed by setting
hidecallerid to permcallerid when calls begin, rather than when they end.
This also means we don't need to also set hidecallerid in chan_dahdi.c
when copying from the config, as we would have to otherwise.
Fourthly, sig_analog currently only allows dialing *67 or *82 if
that would actually toggle the presentation. A comment is added
clarifying that this behavior is okay.
Finally, a couple log messages are updated to be more accurate.
Resolves: #100
ASTERISK-30349 #close
- ### .github: Change title of AsteriskReleaser job
Author: George Joseph
Date: 2023-05-23
- ### asterisk.c: Fix option warning for remote console.
Author: Naveen Albert
Date: 2023-05-18
Commit 09e989f972e2583df4e9bf585c246c37322d8d2f
categorized the T option as not being compatible
with remote consoles, but they do affect verbose
messages with remote console. This fixes this.
Resolves: #102
- ### .github: Don't add cherry-pick reminder if it's already present
Author: George Joseph
Date: 2023-05-22
- ### .github: Fix quoting in PROpenedOrUpdated
Author: George Joseph
Date: 2023-05-16
- ### .github: Add cherry-pick reminder to new PRs
Author: George Joseph
Date: 2023-05-15
- ### configure: fix test code to match gethostbyname_r prototype.
Author: Jaco Kroon
Date: 2023-05-10
This enables the test to work with CC=clang.
Without this the test for 6 args would fail with:
utils.c:99:12: error: static declaration of 'gethostbyname_r' follows non-static declaration
static int gethostbyname_r (const char *name, struct hostent *ret, char *buf,
^
/usr/include/netdb.h:177:12: note: previous declaration is here
extern int gethostbyname_r (const char *__restrict __name,
^
Fixing the expected return type to int sorts this out.
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
- ### res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
Author: Sean Bright
Date: 2023-05-11
The functionality we are interested in is present only in pjsip 2.13
and newer.
Resolves: #45
- ### res_sorcery_memory_cache.c: Fix memory leak
Author: zhengsh
Date: 2023-05-03
Replace the original call to ast_strdup with a call to ast_strdupa to fix the leak issue.
Resolves: #55
ASTERISK-30429
- ### xml.c: Process XML Inclusions recursively.
Author: Sean Bright
Date: 2023-05-09
If processing an XInclude results in new <xi:include> elements, we
need to run XInclude processing again. This continues until no
replacement occurs or an error is encountered.
There is a separate issue with dynamic strings (ast_str) that will be
addressed separately.
Resolves: #65
- ### .github: Tweak improvement issue type language.
Author: Joshua C. Colp
Date: 2023-05-09
- ### .github: Tweak new feature language, and move feature requests elsewhere.
Author: Gitea
Date: 2023-05-09
- ### .github: Fix staleness check to only run on certain labels.
Author: Joshua C. Colp
Date: 2023-05-09

View File

@ -0,0 +1,748 @@
Change Log for Release asterisk-20.5.0
========================================
Links:
----------------------------------------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
Summary:
----------------------------------------
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
User Notes:
----------------------------------------
- ### sig_analog: Add Called Subscriber Held capability.
Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
- ### res_pjsip_header_funcs: Make prefix argument optional.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
- ### core/ari/pjsip: Add refer mechanism
There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.
- ### chan_dahdi: Allow autoreoriginating after hangup.
The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.
- ### sig_analog: Allow three-way flash to time out to silence.
The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
- ### res_pjsip: Enable TLS v1.3 if present.
res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.
- ### app_queue: Add support for applying caller priority change immediately.
The 'queue priority caller' CLI command and
'QueueChangePriorityCaller' AMI action now have an 'immediate'
argument which allows the caller priority change to be reflected
immediately, causing the position of a caller to move within the
queue depending on the priorities of the other callers.
- ### Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
The following manager actions have been added
VoicemailBoxSummary - Generate message list for a given mailbox
VoicemailRemove - Remove a message from a mailbox folder
VoicemailMove - Move a message from one folder to another within a mailbox
VoicemailForward - Copy a message from one folder in one mailbox
to another folder in another or the same mailbox.
- ### app_voicemail: add CLI commands for message manipulation
The following CLI commands have been added to app_voicemail
voicemail show mailbox <mailbox> <context>
Show contents of mailbox <mailbox>@<context>
voicemail remove <mailbox> <context> <from_folder> <messageid>
Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>
voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder>
Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
mailbox <mailbox>@<context> <to_folder>
- ### sig_analog: Allow immediate fake ring to be suppressed.
The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.
Upgrade Notes:
----------------------------------------
Closed Issues:
----------------------------------------
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits
Commits By Author:
----------------------------------------
- ### Asterisk Development Team (1):
- Update for 20.5.0-rc1
- ### Bastian Triller (1):
- res_pjsip_session: Send Session Interval too small response
- ### George Joseph (12):
- .github: Suppress cherry-pick reminder for some situations
- .github: Minor tweak to Asterisk Releaser
- .github: Fix cherry-pick reminder issues
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- rest-api: Run make ari-stubs
- .github: Use generic releaser
- download_externals: Fix a few version related issues
- alembic: Fix quoting of the 100rel column
- .github: Update workflow-application-token-action to v2
- ari-stubs: Fix broken documentation anchors
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ### Holger Hans Peter Freyther (1):
- res_prometheus: Do not generate broken metrics
- ### Jason D. McCormick (1):
- install_prereq: Fix dependency install on aarch64.
- ### Joshua C. Colp (3):
- app_queue: Add support for applying caller priority change immediately.
- audiohook: Unlock channel in mute if no audiohooks present.
- manager: Tolerate stasis messages with no channel snapshot.
- ### Matthew Fredrickson (2):
- Revert "app_stack: Print proper exit location for PBXless channels."
- app_macro: Fix locking around datastore access
- ### Maximilian Fridrich (2):
- core/ari/pjsip: Add refer mechanism
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- ### Mike Bradeen (3):
- app_voicemail: add CLI commands for message manipulation
- Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
- app_voicemail: Fix for loop declarations
- ### MikeNaso (1):
- res_pjsip.c: Set contact_user on incoming call local Contact header
- ### Naveen Albert (7):
- sig_analog: Allow immediate fake ring to be suppressed.
- sig_analog: Allow three-way flash to time out to silence.
- chan_dahdi: Allow autoreoriginating after hangup.
- res_pjsip_header_funcs: Make prefix argument optional.
- sig_analog: Add Called Subscriber Held capability.
- pbx.c: Fix gcc 12 compiler warning.
- app_dial: Fix infinite loop when sending digits.
- ### Sean Bright (6):
- res_geolocation: Ensure required 'location_info' is present.
- chan_iax2.c: Avoid crash with IAX2 switch support.
- func_export: Use correct function argument as variable name.
- extensions.conf.sample: Remove reference to missing context.
- res_pjsip: Enable TLS v1.3 if present.
- extconfig: Allow explicit DB result set ordering to be disabled.
- ### phoneben (1):
- func_cut: Add example to documentation.
- ### zhengsh (2):
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
Detail:
----------------------------------------
- ### ari-stubs: Fix more local anchor references
Author: George Joseph
Date: 2023-09-05
Also allow CreateDocs job to be run manually with default branches.
- ### ari-stubs: Fix more local anchor references
Author: George Joseph
Date: 2023-09-05
Also allow CreateDocs job to be run manually with default branches.
- ### ari-stubs: Fix broken documentation anchors
Author: George Joseph
Date: 2023-09-05
All of the links that reference page anchors with capital letters in
the ids (#Something) have been changed to lower case to match the
anchors that are generated by mkdocs.
- ### res_pjsip_session: Send Session Interval too small response
Author: Bastian Triller
Date: 2023-08-28
Handle session interval lower than endpoint's configured minimum timer
when sending first answer. Timer setting is checked during this step and
needs to handled appropriately.
Before this change, no response was sent at all. After this change a
response with 422 Session Interval too small is sent to UAC.
- ### .github: Update workflow-application-token-action to v2
Author: George Joseph
Date: 2023-08-31
- ### app_dial: Fix infinite loop when sending digits.
Author: Naveen Albert
Date: 2023-08-28
If the called party hangs up while digits are being
sent, -1 is returned to indicate so, but app_dial
was not checking the return value, resulting in
the hangup being lost and looping forever until
the caller manually hangs up the channel. We now
abort if digit sending fails.
ASTERISK-29428 #close
Resolves: #281
- ### app_voicemail: Fix for loop declarations
Author: Mike Bradeen
Date: 2023-08-29
Resolve for loop initial declarations added in cli changes.
Resolves: #275
- ### alembic: Fix quoting of the 100rel column
Author: George Joseph
Date: 2023-08-28
Add quoting around the ps_endpoints 100rel column in the ALTER
statements. Although alembic doesn't complain when generating
sql statements, postgresql does (rightly so).
Resolves: #274
- ### pbx.c: Fix gcc 12 compiler warning.
Author: Naveen Albert
Date: 2023-08-27
Resolves: #277
- ### app_audiosocket: Fixed timeout with -1 to avoid busy loop.
Author: zhengsh
Date: 2023-08-24
Resolves: asterisk#234
- ### download_externals: Fix a few version related issues
Author: George Joseph
Date: 2023-08-18
* Fixed issue with the script not parsing the new tag format for
certified releases. The format changed from certified/18.9-cert5
to certified-18.9-cert5.
* Fixed issue where the asterisk version wasn't being considered
when looking for cached versions.
Resolves: #263
- ### main/refer.c: Fix double free in refer_data_destructor + potential leak
Author: Maximilian Fridrich
Date: 2023-08-21
Resolves: #267
- ### sig_analog: Add Called Subscriber Held capability.
Author: Naveen Albert
Date: 2023-08-09
This adds support for Called Subscriber Held for FXS
lines, which allows users to go on hook when receiving
a call and resume the call later from another phone on
the same line, without disconnecting the call. This is
a convenience mechanism that most real PSTN telephone
switches support.
ASTERISK-30372 #close
Resolves: #240
UserNote: Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
- ### app_macro: Fix locking around datastore access
Author: Matthew Fredrickson
Date: 2023-08-21
app_macro sometimes would crash due to datastore list corruption on the
channel because of lack of locking around find and create process for
the macro datastore. This patch locks the channel lock prior to protect
against this problem.
Resolves: #265
- ### Revert "app_stack: Print proper exit location for PBXless channels."
Author: Matthew Fredrickson
Date: 2023-08-10
This reverts commit 617dad4cba1513dddce87b8e95a61415fb587cf1.
apps/app_stack.c: Revert buggy gosub patch
This seems to break the case when a predial macro calls a gosub.
When the gosub calls return, the Return function outputs:
app_stack.c:423 return_exec: Return without Gosub: stack is empty
This returns -1 to the calling macro, which returns to app_dial
and causes the call to hangup instead of proceeding with the macro
that invoked the gosub.
Resolves: #253
- ### .github: Use generic releaser
Author: George Joseph
Date: 2023-08-15
- ### install_prereq: Fix dependency install on aarch64.
Author: Jason D. McCormick
Date: 2023-04-28
Fixes dependency solutions in install_prereq for Debian aarch64
platforms. install_prereq was attempting to forcibly install 32-bit
armhf packages due to the aptitude search for dependencies.
Resolves: #37
- ### res_pjsip.c: Set contact_user on incoming call local Contact header
Author: MikeNaso
Date: 2023-08-08
If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.
Resolves: #226
- ### extconfig: Allow explicit DB result set ordering to be disabled.
Author: Sean Bright
Date: 2023-07-12
Added a new boolean configuration flag -
`order_multi_row_results_by_initial_column` - to both res_pgsql.conf
and res_config_odbc.conf that allows the administrator to disable the
explicit `ORDER BY` that was previously being added to all generated
SQL statements that returned multiple rows.
Fixes: #179
- ### rest-api: Run make ari-stubs
Author: George Joseph
Date: 2023-08-09
An earlier cherry-pick that involved rest-api somehow didn't include
a comment change in res/ari/resource_endpoints.h. This commit
corrects that. No changes other than the comment.
- ### res_pjsip_header_funcs: Make prefix argument optional.
Author: Naveen Albert
Date: 2023-08-09
The documentation for PJSIP_HEADERS claims that
prefix is optional, but in the code it is actually not.
However, there is no inherent reason for this, as users
may want to retrieve all header names, not just those
beginning with a certain prefix.
This makes the prefix optional for this function,
simply fetching all header names if not specified.
As a result, the documentation is now correct.
Resolves: #230
UserNote: The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
- ### pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
Author: George Joseph
Date: 2023-08-11
The default is 32 with 8 being used by pjproject itself. Recent
commits have put us over the limit resulting in assertions in
pjproject. Since this value is used in invites, dialogs,
transports and subscriptions as well as the global pjproject
endpoint, we don't want to increase it too much.
Resolves: #255
- ### manager: Tolerate stasis messages with no channel snapshot.
Author: Joshua C. Colp
Date: 2023-08-09
In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.
- ### core/ari/pjsip: Add refer mechanism
Author: Maximilian Fridrich
Date: 2023-05-10
This change adds support for refers that are not session based. It
includes a refer implementation for the PJSIP technology which results
in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
triggered using the new ARI endpoint `/endpoints/refer`.
Resolves: #71
UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.
- ### chan_dahdi: Allow autoreoriginating after hangup.
Author: Naveen Albert
Date: 2023-08-04
Currently, if an FXS channel is still off hook when
all calls on the line have hung up, the user is provided
reorder tone until going back on hook again.
In addition to not reflecting what most commercial switches
actually do, it's very common for switches to automatically
reoriginate for the user so that dial tone is provided without
the user having to depress and release the hookswitch manually.
This can increase convenience for users.
This behavior is now supported for kewlstart FXS channels.
It's supported only for kewlstart (FXOKS) mainly because the
behavior doesn't make any sense for ground start channels,
and loop start signalling doesn't provide the necessary DAHDI
event that makes this easy to implement. Likely almost everyone
is using FXOKS over FXOLS anyways since FXOLS is pretty useless
these days.
ASTERISK-30357 #close
Resolves: #224
UserNote: The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.
- ### audiohook: Unlock channel in mute if no audiohooks present.
Author: Joshua C. Colp
Date: 2023-08-09
In the case where mute was called on a channel that had no
audiohooks the code was not unlocking the channel, resulting
in a deadlock.
Resolves: #233
- ### sig_analog: Allow three-way flash to time out to silence.
Author: Naveen Albert
Date: 2023-07-10
sig_analog allows users to flash and use the three-way dial
tone as a primitive hold function, simply by never timing
it out.
Some systems allow this dial tone to time out to silence,
so the user is not annoyed by a persistent dial tone.
This option allows the dial tone to time out normally to
silence.
ASTERISK-30004 #close
Resolves: #205
UserNote: The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
- ### res_prometheus: Do not generate broken metrics
Author: Holger Hans Peter Freyther
Date: 2023-04-07
In 8d6fdf9c3adede201f0ef026dab201b3a37b26b6 invisible bridges were
skipped but that lead to producing metrics with no name and no help.
Keep track of the number of metrics configured and then only emit these.
Add a basic testcase that verifies that there is no '(NULL)' in the
output.
ASTERISK-30474
- ### res_pjsip: Enable TLS v1.3 if present.
Author: Sean Bright
Date: 2023-08-02
Fixes #221
UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.
- ### func_cut: Add example to documentation.
Author: phoneben
Date: 2023-07-19
This adds an example to the XML documentation clarifying usage
of the CUT function to address a common misusage.
- ### extensions.conf.sample: Remove reference to missing context.
Author: Sean Bright
Date: 2023-07-16
c3ff4648 removed the [iaxtel700] context but neglected to remove
references to it.
This commit addresses that and also removes iaxtel and freeworlddialup
references from other config files.
- ### func_export: Use correct function argument as variable name.
Author: Sean Bright
Date: 2023-07-12
Fixes #208
- ### app_queue: Add support for applying caller priority change immediately.
Author: Joshua C. Colp
Date: 2023-07-07
The app_queue module provides both an AMI action and a CLI command
to change the priority of a caller in a queue. Up to now this change
of priority has only been reflected to new callers into the queue.
This change adds an "immediate" option to both the AMI action and
CLI command which immediately applies the priority change respective
to the other callers already in the queue. This can allow, for example,
a caller to be placed at the head of the queue immediately if their
priority is sufficient.
Resolves: #202
UserNote: The 'queue priority caller' CLI command and
'QueueChangePriorityCaller' AMI action now have an 'immediate'
argument which allows the caller priority change to be reflected
immediately, causing the position of a caller to move within the
queue depending on the priorities of the other callers.
- ### .github: Fix cherry-pick reminder issues
Author: George Joseph
Date: 2023-07-17
- ### chan_iax2.c: Avoid crash with IAX2 switch support.
Author: Sean Bright
Date: 2023-07-07
A change made in 82cebaa0 did not properly handle the case when a
channel was not provided, triggering a crash. ast_check_hangup(...)
does not protect against NULL pointers.
Fixes #180
- ### res_geolocation: Ensure required 'location_info' is present.
Author: Sean Bright
Date: 2023-07-07
Fixes #189
- ### Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
Author: Mike Bradeen
Date: 2023-06-29
Resolves: #181
UserNote: The following manager actions have been added
VoicemailBoxSummary - Generate message list for a given mailbox
VoicemailRemove - Remove a message from a mailbox folder
VoicemailMove - Move a message from one folder to another within a mailbox
VoicemailForward - Copy a message from one folder in one mailbox
to another folder in another or the same mailbox.
- ### app_voicemail: add CLI commands for message manipulation
Author: Mike Bradeen
Date: 2023-06-20
Adds CLI commands to allow move/remove/forward individual messages
from a particular mailbox folder. The forward command can be used
to copy a message within a mailbox or to another mailbox. Also adds
a show mailbox, required to retrieve message ID's.
Resolves: #170
UserNote: The following CLI commands have been added to app_voicemail
voicemail show mailbox <mailbox> <context>
Show contents of mailbox <mailbox>@<context>
voicemail remove <mailbox> <context> <from_folder> <messageid>
Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>
voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder>
Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
mailbox <mailbox>@<context> <to_folder>
- ### res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
Author: zhengsh
Date: 2023-06-30
From the gdb information, it was found that when calling __ast_free, the size of the
allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid
is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb,
it is found to be 1.
Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid,
which is outside the protection of the rtp_instance lock. However,
ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses
rtp->themssrc_valid within the protection of the rtp_instance lock.
This can lead to the possibility that the value of rtp->themssrc_valid used in the call to
ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used
within ast_rtcp_generate_report().
Resolves: asterisk#63
- ### .github: Minor tweak to Asterisk Releaser
Author: George Joseph
Date: 2023-07-12
- ### .github: Suppress cherry-pick reminder for some situations
Author: George Joseph
Date: 2023-07-11
In PROpenedOrUpdated, the cherry-pick reminder will now be
suppressed if there are already valid 'cherry-pick-to' comments
in the PR or the PR contained a 'cherry-pick-to: none' comment.
- ### sig_analog: Allow immediate fake ring to be suppressed.
Author: Naveen Albert
Date: 2023-06-08
When immediate=yes on an FXS channel, sig_analog will
start fake audible ringback that continues until the
channel is answered. Even if it answers immediately,
the ringback is still audible for a brief moment.
This can be disruptive and unwanted behavior.
This adds an option to disable this behavior, though
the default behavior remains unchanged.
ASTERISK-30003 #close
Resolves: #118
UserNote: The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.

View File

@ -0,0 +1,95 @@
Change Log for Release asterisk-20.5.1
========================================
Links:
----------------------------------------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
Summary:
----------------------------------------
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
User Notes:
----------------------------------------
Upgrade Notes:
----------------------------------------
Closed Issues:
----------------------------------------
None
Commits By Author:
----------------------------------------
- ### Ben Ford (1):
- manager.c: Prevent path traversal with GetConfig.
- ### George Joseph (1):
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- ### Gitea (1):
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- ### Mike Bradeen (1):
- res_pjsip: disable raw bad packet logging
Detail:
----------------------------------------
- ### res_pjsip_header_funcs: Duplicate new header value, don't copy.
Author: Gitea
Date: 2023-07-10
When updating an existing header the 'update' code incorrectly
just copied the new value into the existing buffer. If the
new value exceeded the available buffer size memory outside
of the buffer would be written into, potentially causing
a crash.
This change makes it so that the 'update' now duplicates
the new header value instead of copying it into the existing
buffer.
- ### res_pjsip: disable raw bad packet logging
Author: Mike Bradeen
Date: 2023-07-25
Add patch to split the log level for invalid packets received on the
signaling port. The warning regarding the packet will move to level 2
so that it can still be displayed, while the raw packet will be at level
4.
- ### res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
Author: George Joseph
Date: 2023-11-09
When ICE is in use, we can prevent a possible DOS attack by allowing
DTLS protocol messages (client hello, etc) only from sources that
are in the active remote candidates list.
Resolves: GHSA-hxj9-xwr8-w8pq
- ### manager.c: Prevent path traversal with GetConfig.
Author: Ben Ford
Date: 2023-11-13
When using AMI GetConfig, it was possible to access files outside of the
Asterisk configuration directory by using filenames with ".." and "./"
even while live_dangerously was not enabled. This change resolves the
full path and ensures we are still in the configuration directory before
attempting to access the file.

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@ -0,0 +1,68 @@
Change Log for Release asterisk-20.5.2
========================================
Links:
----------------------------------------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
Summary:
----------------------------------------
- res_rtp_asterisk: Fix regression issues with DTLS client check
User Notes:
----------------------------------------
Upgrade Notes:
----------------------------------------
Closed Issues:
----------------------------------------
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()
Commits By Author:
----------------------------------------
- ### George Joseph (1):
- res_rtp_asterisk: Fix regression issues with DTLS client check
Detail:
----------------------------------------
- ### res_rtp_asterisk: Fix regression issues with DTLS client check
Author: George Joseph
Date: 2023-12-15
* Since ICE candidates are used for the check and pjproject is
required to use ICE, res_rtp_asterisk was failing to compile
when pjproject wasn't available. The check is now wrapped
with an #ifdef HAVE_PJPROJECT.
* The rtp->ice_active_remote_candidates container was being
used to check the address on incoming packets but that
container doesn't contain peer reflexive candidates discovered
during negotiation. This was causing the check to fail
where it shouldn't. We now check against pjproject's
real_ice->rcand array which will contain those candidates.
* Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where
we weren't zeroing out sin->sin_zero before returning. This
was causing ast_sockaddr_cmp() to always return false when
one of the inputs was converted from a pj_sockaddr, even
if both inputs had the same address and port.
Resolves: #500
Resolves: #503
Resolves: #505

File diff suppressed because it is too large Load Diff

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@ -0,0 +1,736 @@
Change Log for Release asterisk-20.7.0
========================================
Links:
----------------------------------------
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
Summary:
----------------------------------------
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
User Notes:
----------------------------------------
- ### app_dial: Add dial time for progress/ringing.
The timeout argument to Dial now allows
specifying the maximum amount of time to dial if
early media is not received.
- ### app_voicemail: Allow preventing mark messages as urgent.
The leaveurgent mailbox option can now be used to
control whether callers may leave messages marked as 'Urgent'.
- ### Stir/Shaken Refactor
Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.
- ### Upgrade bundled pjproject to 2.14.
Bundled pjproject has been upgraded to 2.14. For more
information on what all is included in this change, check out the
pjproject Github page: https://github.com/pjsip/pjproject/releases
- ### app_speech_utils.c: Allow partial speech results.
The SpeechBackground dialplan application now supports a 'p'
option that will return partial results from speech engines that
provide them when a timeout occurs.
- ### app_chanspy: Add 'D' option for dual-channel audio
The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
The 'dahdi set mwi' now allows MWI on channels
to be manually toggled if needed for troubleshooting.
Resolves: #440
Upgrade Notes:
----------------------------------------
- ### Stir/Shaken Refactor
The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed. The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information. This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added. Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.
Closed Issues:
----------------------------------------
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c
Commits By Author:
----------------------------------------
- ### Ben Ford (1):
- Upgrade bundled pjproject to 2.14.
- ### Brad Smith (2):
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- BuildSystem: Bump autotools versions on OpenBSD.
- ### George Joseph (6):
- Reduce startup/shutdown verbose logging
- pjsip show channelstats: Prevent possible segfault when faxing
- Stir/Shaken Refactor
- Makefile: Add stir_shaken/cache to directories created on install
- attestation_config.c: Use ast_free instead of ast_std_free
- res_pjsip_stir_shaken.c: Add checks for missing parameters
- ### Joshua C. Colp (1):
- utils: Make behavior of ast_strsep* match strsep.
- ### Mike Bradeen (2):
- app_voicemail: add NoOp alembic script to maintain sync
- app_chanspy: Add 'D' option for dual-channel audio
- ### Naveen Albert (10):
- func_frame_trace: Add CLI command to dump frame queue.
- chan_dahdi: Allow MWI to be manually toggled on channels.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- manager.c: Fix erroneous reloads in UpdateConfig.
- app_if: Fix next priority calculation.
- configure: Rerun bootstrap on modern platform.
- dsp.c: Fix and improve potentially inaccurate log message.
- app_voicemail: Allow preventing mark messages as urgent.
- app_voicemail: Properly reinitialize config after unit tests.
- app_dial: Add dial time for progress/ringing.
- ### PeterHolik (2):
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- chan_rtp.c: MulticastRTP missing refcount without codec option
- ### Sean Bright (7):
- app_confbridge: Don't emit warnings on valid configurations.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- res_pjsip_session.c: Correctly format SDP connection addresses.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- alembic: Synchronize alembic heads between supported branches.
- res_pjsip: Use consistent type for boolean columns.
- ### Sebastian Jennen (1):
- translate.c: implement new direct comp table mode
- ### Shaaah (1):
- app_queue.c : fix "queue add member" usage string
- ### Shyju Kanaprath (1):
- README.md: Removed outdated link
- ### cmaj (1):
- app_speech_utils.c: Allow partial speech results.
- ### romryz (1):
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
Detail:
----------------------------------------
- ### res_pjsip_stir_shaken.c: Add checks for missing parameters
Author: George Joseph
Date: 2024-03-11
* Added checks for missing session, session->channel and rdata
in stir_shaken_incoming_request.
* Added checks for missing session, session->channel and tdata
in stir_shaken_outgoing_request.
Resolves: #645
- ### app_dial: Add dial time for progress/ringing.
Author: Naveen Albert
Date: 2024-02-08
Add a timeout option to control the amount of time
to wait if no early media is received before giving
up. This allows aborting early if the destination
is not being responsive.
Resolves: #588
UserNote: The timeout argument to Dial now allows
specifying the maximum amount of time to dial if
early media is not received.
- ### app_voicemail: Properly reinitialize config after unit tests.
Author: Naveen Albert
Date: 2024-02-29
Most app_voicemail unit tests were not properly cleaning up
after themselves after running. This led to test mailboxes
lingering around in the system. It also meant that if any
unit tests in app_voicemail that create mailboxes were executed
and the module was not unloaded/loaded again prior to running
the test_voicemail_vm_info unit test, Asterisk would segfault
due to an attempt to copy a NULL string.
The load_config test did actually have logic to reinitialize
the config after the test. However, this did not work in practice
since load_config() would not reload the config since voicemail.conf
had not changed during the test; thus, additional logic has been
added to ensure that voicemail.conf is truly reloaded, after any
unit tests which modify the users list.
This prevents the SEGV due to invalid mailboxes lingering around,
and also ensures that the system state is restored to what it was
prior to the tests running.
Resolves: #629
- ### app_queue.c : fix "queue add member" usage string
Author: Shaaah
Date: 2024-01-23
Fixing bracket placement in the "queue add member" cli usage string.
- ### app_voicemail: Allow preventing mark messages as urgent.
Author: Naveen Albert
Date: 2024-02-24
This adds an option to allow preventing callers from leaving
messages marked as 'urgent'.
Resolves: #619
UserNote: The leaveurgent mailbox option can now be used to
control whether callers may leave messages marked as 'Urgent'.
- ### res_pjsip: Use consistent type for boolean columns.
Author: Sean Bright
Date: 2024-02-27
This migrates the relevant schema objects from the `('yes', 'no')`
definition to the `('0', '1', 'off', 'on', 'false', 'true', 'yes', 'no')`
one.
Fixes #617
- ### attestation_config.c: Use ast_free instead of ast_std_free
Author: George Joseph
Date: 2024-03-05
In as_check_common_config, we were calling ast_std_free on
raw_key but raw_key was allocated with ast_malloc so it
should be freed with ast_free.
Resolves: #636
- ### Makefile: Add stir_shaken/cache to directories created on install
Author: George Joseph
Date: 2024-03-04
The default location for the stir_shaken cache is
/var/lib/asterisk/keys/stir_shaken/cache but we were only creating
/var/lib/asterisk/keys/stir_shaken on istall. We now create
the cache sub-directory.
Resolves: #634
- ### Stir/Shaken Refactor
Author: George Joseph
Date: 2023-10-26
Why do we need a refactor?
The original stir/shaken implementation was started over 3 years ago
when little was understood about practical implementation. The
result was an implementation that wouldn't actually interoperate
with any other stir-shaken implementations.
There were also a number of stir-shaken features and RFC
requirements that were never implemented such as TNAuthList
certificate validation, sending Reason headers in SIP responses
when verification failed but we wished to continue the call, and
the ability to send Media Key(mky) grants in the Identity header
when the call involved DTLS.
Finally, there were some performance concerns around outgoing
calls and selection of the correct certificate and private key.
The configuration was keyed by an arbitrary name which meant that
for every outgoing call, we had to scan the entire list of
configured TNs to find the correct cert to use. With only a few
TNs configured, this wasn't an issue but if you have a thousand,
it could be.
What's changed?
* Configuration objects have been refactored to be clearer about
their uses and to fix issues.
* The "general" object was renamed to "verification" since it
contains parameters specific to the incoming verification
process. It also never handled ca_path and crl_path
correctly.
* A new "attestation" object was added that controls the
outgoing attestation process. It sets default certificates,
keys, etc.
* The "certificate" object was renamed to "tn" and had it's key
change to telephone number since outgoing call attestation
needs to look up certificates by telephone number.
* The "profile" object had more parameters added to it that can
override default parameters specified in the "attestation"
and "verification" objects.
* The "store" object was removed altogther as it was never
implemented.
* We now use libjwt to create outgoing Identity headers and to
parse and validate signatures on incoming Identiy headers. Our
previous custom implementation was much of the source of the
interoperability issues.
* General code cleanup and refactor.
* Moved things to better places.
* Separated some of the complex functions to smaller ones.
* Using context objects rather than passing tons of parameters
in function calls.
* Removed some complexity and unneeded encapsuation from the
config objects.
Resolves: #351
Resolves: #46
UserNote: Asterisk's stir-shaken feature has been refactored to
correct interoperability, RFC compliance, and performance issues.
See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
information.
UpgradeNote: The stir-shaken refactor is a breaking change but since
it's not working now we don't think it matters. The
stir_shaken.conf file has changed significantly which means that
existing ones WILL need to be changed. The stir_shaken.conf.sample
file in configs/samples/ has quite a bit more information. This is
also an ABI breaking change since some of the existing objects
needed to be changed or removed, and new ones added. Additionally,
if res_stir_shaken is enabled in menuselect, you'll need to either
have the development package for libjwt v1.15.3 installed or use
the --with-libjwt-bundled option with ./configure.
- ### alembic: Synchronize alembic heads between supported branches.
Author: Sean Bright
Date: 2024-02-28
This adds a dummy migration to 18 and 20 so that our alembic heads are
synchronized across all supported branches.
In this case the migration we are stubbing (24c12d8e9014) is:
https://github.com/asterisk/asterisk/commit/775352ee6c2a5bcd4f0e3df51aee5d1b0abf4cbe
- ### translate.c: implement new direct comp table mode
Author: Sebastian Jennen
Date: 2024-02-25
The new mode lists for each codec translation the actual real cost in cpu microseconds per second translated audio.
This allows to compare the real cpu usage of translations and helps in evaluation of codec implementation changes regarding performance (regression testing).
- add new table mode
- hide the 999999 comp values, as these only indicate an issue with transcoding
- hide the 0 values, as these also do not contain any information (only indicate a multistep transcoding)
Resolves: #601
- ### README.md: Removed outdated link
Author: Shyju Kanaprath
Date: 2024-02-23
Removed outdated link http://www.quicknet.net from README.md
cherry-pick-to: 18
cherry-pick-to: 20
cherry-pick-to: 21
- ### strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
Author: Sean Bright
Date: 2024-02-17
If a dynamic string is created with an initial length of 0,
`ast_str_buffer(…)` will return an invalid pointer.
This was a secondary discovery when fixing #65.
- ### res_rtp_asterisk.c: Correct coefficient in MOS calculation.
Author: romryz
Date: 2024-02-06
Media Experience Score relies on incorrect pseudo_mos variable
calculation. According to forming an opinion section of the
documentation, calculation relies on ITU-T G.107 standard:
https://docs.asterisk.org/Deployment/Media-Experience-Score/#forming-an-opinion
ITU-T G.107 Annex B suggests to calculate MOS with a coefficient
"seven times ten to the power of negative six", 7 * 10^(-6). which
would mean 6 digits after the decimal point. Current implementation
has 7 digits after the decimal point, which downrates the calls.
Fixes: #597
- ### dsp.c: Fix and improve potentially inaccurate log message.
Author: Naveen Albert
Date: 2024-02-09
If ast_dsp_process is called with a codec besides slin, ulaw,
or alaw, a warning is logged that in-band DTMF is not supported,
but this message is not always appropriate or correct, because
ast_dsp_process is much more generic than just DTMF detection.
This logs a more generic message in those cases, and also improves
codec-mismatch logging throughout dsp.c by ensuring incompatible
codecs are printed out.
Resolves: #595
- ### pjsip show channelstats: Prevent possible segfault when faxing
Author: George Joseph
Date: 2024-02-09
Under rare circumstances, it's possible for the original audio
session in the active_media_state default_session to be corrupted
instead of removed when switching to the t38/image media session
during fax negotiation. This can cause a segfault when a "pjsip
show channelstats" attempts to print that audio media session's
rtp statistics. In these cases, the active_media_state
topology is correctly showing only a single t38/image stream
so we now check that there's an audio stream in the topology
before attempting to use the audio media session to get the rtp
statistics.
Resolves: #592
- ### Reduce startup/shutdown verbose logging
Author: George Joseph
Date: 2024-01-31
When started with a verbose level of 3, asterisk can emit over 1500
verbose message that serve no real purpose other than to fill up
logs. When asterisk shuts down, it emits another 1100 that are of
even less use. Since the testsuite runs asterisk with a verbose
level of 3, and asterisk starts and stops for every one of the 700+
tests, the number of log messages is staggering. Besides taking up
resources, it also makes it hard to debug failing tests.
This commit changes the log level for those verbose messages to 5
instead of 3 which reduces the number of log messages to only a
handful. Of course, NOTICE, WARNING and ERROR message are
unaffected.
There's also one other minor change...
ast_context_remove_extension_callerid2() logs a DEBUG message
instead of an ERROR if the extension you're deleting doesn't exist.
The pjsip_config_wizard calls that function to clean up the config
and has been triggering that annoying error message for years.
Resolves: #582
- ### configure: Rerun bootstrap on modern platform.
Author: Naveen Albert
Date: 2024-02-12
The last time configure was run, it was run on a system that
did not enable -std=gnu11 by default, which meant that the
restrict qualifier would not be recognized on certain platforms.
This regenerates the configure files from running bootstrap.sh,
so that these should be recognized on all supported platforms.
Resolves: #586
- ### Upgrade bundled pjproject to 2.14.
Author: Ben Ford
Date: 2024-02-05
Fixes: #406
UserNote: Bundled pjproject has been upgraded to 2.14. For more
information on what all is included in this change, check out the
pjproject Github page: https://github.com/pjsip/pjproject/releases
- ### app_speech_utils.c: Allow partial speech results.
Author: cmaj
Date: 2024-02-02
Adds 'p' option to SpeechBackground() application.
With this option, when the app timeout is reached,
whatever the backend speech engine collected will
be returned as if it were the final, full result.
(This works for engines that make partial results.)
Resolves: #572
UserNote: The SpeechBackground dialplan application now supports a 'p'
option that will return partial results from speech engines that
provide them when a timeout occurs.
- ### utils: Make behavior of ast_strsep* match strsep.
Author: Joshua C. Colp
Date: 2024-01-31
Given the scenario of passing an empty string to the
ast_strsep functions the functions would return NULL
instead of an empty string. This is counter to how
strsep itself works.
This change alters the behavior of the functions to
match that of strsep.
Fixes: #565
- ### app_chanspy: Add 'D' option for dual-channel audio
Author: Mike Bradeen
Date: 2024-01-31
Adds the 'D' option to app chanspy that causes the input and output
frames of the spied channel to be interleaved in the spy output frame.
This allows the input and output of the spied channel to be decoded
separately by the receiver.
If the 'o' option is also set, the 'D' option is ignored as the
audio being spied is inherently one direction.
Fixes: #569
UserNote: The ChanSpy application now accepts the 'D' option which
will interleave the spied audio within the outgoing frames. The
purpose of this is to allow the audio to be read as a Dual channel
stream with separate incoming and outgoing audio. Setting both the
'o' option and the 'D' option and results in the 'D' option being
ignored.
- ### app_if: Fix next priority calculation.
Author: Naveen Albert
Date: 2024-01-28
Commit fa3922a4d28860d415614347d9f06c233d2beb07 fixed
a branching issue but "overshoots" when calculating
the next priority. This fixes that; accompanying
test suite tests have also been extended.
Resolves: #560
- ### res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
Author: Sean Bright
Date: 2024-01-29
The existing code prevented IPv6 addresses from being properly parsed.
Fixes #558
- ### BuildSystem: Bump autotools versions on OpenBSD.
Author: Brad Smith
Date: 2024-01-27
Bump up to the more commonly used and modern versions of
autoconf and automake.
- ### main/utils: Simplify the FreeBSD ast_get_tid() handling
Author: Brad Smith
Date: 2024-01-27
FreeBSD has had kernel threads for 20+ years.
- ### res_pjsip_session.c: Correctly format SDP connection addresses.
Author: Sean Bright
Date: 2024-01-27
Resolves a regression identified by @justinludwig involving the
rendering of IPv6 addresses in outgoing SDP.
Also updates `media_address` on PJSIP endpoints so that if we are able
to parse the configured value as an IP we store it in a format that we
can directly use later. Based on my reading of the code it appeared
that one could configure `media_address` as:
```
[foo]
type = endpoint
...
media_address = [2001:db8::]
```
And that value would be blindly copied into the outgoing SDP without
regard to its format.
Fixes #541
- ### rtp_engine.c: Correct sample rate typo for L16/44100.
Author: Sean Bright
Date: 2024-01-28
Fixes #555
- ### manager.c: Fix erroneous reloads in UpdateConfig.
Author: Naveen Albert
Date: 2024-01-25
Currently, a reload will always occur if the
Reload header is provided for the UpdateConfig
action. However, we should not be doing a reload
if the header value has a falsy value, per the
documentation, so this makes the reload behavior
consistent with the existing documentation.
Resolves: #551
- ### res_calendar_icalendar: Print iCalendar error on parsing failure.
Author: Naveen Albert
Date: 2023-12-14
If libical fails to parse a calendar, print the error message it provdes.
Resolves: #492
- ### app_confbridge: Don't emit warnings on valid configurations.
Author: Sean Bright
Date: 2024-01-21
The numeric bridge profile options `internal_sample_rate` and
`maximum_sample_rate` are documented to accept the special values
`auto` and `none`, respectively. While these values currently work,
they also emit warnings when used which could be confusing for users.
In passing, also ensure that we only accept the documented range of
sample rate values between 8000 and 192000.
Fixes #546
- ### app_voicemail: add NoOp alembic script to maintain sync
Author: Mike Bradeen
Date: 2024-01-17
Adding a NoOp alembic script for the voicemail database to maintain
version sync with other branches.
Fixes: #527
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
Author: Naveen Albert
Date: 2023-11-10
This adds a CLI command to manually toggle the MWI status
of a channel, useful for troubleshooting or resetting
MWI devices, similar to the capabilities offered with
SIP messaging to manually control MWI status.
UserNote: The 'dahdi set mwi' now allows MWI on channels
to be manually toggled if needed for troubleshooting.
Resolves: #440
- ### chan_rtp.c: MulticastRTP missing refcount without codec option
Author: PeterHolik
Date: 2024-01-15
Fixes: #529
- ### chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
Author: PeterHolik
Date: 2024-01-16
Fixes: asterisk#536
- ### func_frame_trace: Add CLI command to dump frame queue.
Author: Naveen Albert
Date: 2024-01-12
This adds a simple CLI command that can be used for
analyzing all frames currently queued to a channel.
A couple log messages are also adjusted to be more
useful in tracing bridging problems.
Resolves: #533

View File

@ -12,10 +12,316 @@
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 20.1.0 to Asterisk 20.2.0 ------------
------------------------------------------------------------------------------
app_broadcast
------------------
* A Broadcast application is now available which allows
for asynchronous one-to-many and many-to-one channel audio.
app_directory
------------------
* A new option 's' has been added to the Directory() application that
will skip calling the extension and instead set the extension as
DIRECTORY_EXTEN channel variable.
app_read
------------------
* A new option 'e' has been added to allow Read() to return the
terminator as the dialed digits in the case where only the terminator
is entered.
app_senddtmf
------------------
* A new option has been added to SendDTMF() which will answer the
specified channel if it is not already up. If no channel is specified,
the current channel will be answered instead.
app_signal
------------------
* Adds Signal and WaitForSignal applications
which can be used for signaling or as a
simple message queue in the dialplan.
func_json
------------------
* Additional parsing capabilities have been added to the
JSON_DECODE function, including support for arrays
and recursive indexing.
res_phoneprov
------------------
* On multihomed Asterisk servers with dynamic SERVER template variables,
reloading this module is no longer required when re-provisioning your
phone to another interface address (e.g. when moving between VLANs.)
res_pjsip_rfc3326
------------------
* Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in
addition to currently supported Q.850). The first header found will be used to set
the HANGUPCAUSE variable.
res_pjsip_session
------------------
* The overlap_context option now allows explicitly
specifying a context to use for overlap dialing matches.
res_rtp_asterisk
------------------
* This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score. The score is available using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics. For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------
------------------------------------------------------------------------------
AMI
------------------
* The AOCMessage action can now be used to generate AOC-S messages.
Add support for named capture agent.
------------------
* A name for the capture agent can now be specified
using the capture_name option which, if specified,
will be sent to the HEP server.
app_if
------------------
* Adds the If, ElseIf, Else, EndIf, and ExitIf applications
for conditional execution of a block of code.
app_mixmonitor
------------------
* The d option for MixMonitor now allows deleting
the original recording when MixMonitor exits,
which can be useful when MixMonitor copies it
somewhere else before exiting.
* Adds the c option to use the real Caller ID on
the channel in voicemail recordings as opposed
to the Connected Line.
app_voicemail
------------------
* The voicemail user option attachextrecs can
now be set to control whether external recordings
trigger voicemail email notifications.
cdr
------------------
* Two new options have been added which allow
bridging and dial state changes to be ignored
in CDRs, which can be useful if a single CDR
is desired for a channel.
chan_dahdi
------------------
* FXO channels (FXS signaled) that don't use callerid or
distinctive ring detection can now be configured
to enter the dialplan immediately using immediate=yes,
instead of waiting for at least one ring.
pbx_builtins
------------------
* It is now possible to not wait for media on
a channel when answering it using Answer,
by specifying the i option.
res_pjsip
------------------
* Added options "security_negotiation" and "security_mechanisms" to pjsip
endpoints and registrations. "security_negotiation" can be set to "no" (default)
or "mediasec", and "security_mechanisms" can be a list of comma-separated
security_mechanisms in the form defined by RFC 3329 section 2.2.
* A new option named "all_codecs_on_empty_reinvite" has been added to the
global section. When this option is enabled, on reception of a re-INVITE
without SDP, Asterisk will send an SDP offer in the 200 OK response containing
all configured codecs on the endpoint, instead of simply those that have
already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
The default value is "off".
res_pjsip_aoc
------------------
* Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages.
A new endpoint option, send_aoc, controls this.
res_pjsip_header_funcs
------------------
* The new PJSIP_HEADER_PARAM function now fully supports both
URI and header parameters. Both reading and writing
parameters are supported.
res_pjsip_logger
------------------
* SIP messages can now be filtered by SIP request method
(INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
allowing for more granular debugging to be done
in the CLI. This applies to requests but not responses.
res_pjsip_notify
------------------
* Allows using the config options in pjsip_notify.conf
from AMI actions as with the existing CLI commands.
res_tonedetect
------------------
* The TONE_DETECT function now supports
detection of audible ringback tone
using the p option.
xmldocs
------------------
* The XML documentation can now be reloaded without restarting
Asterisk, which makes it possible to load new modules that
enforce documentation without restarting Asterisk.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
------------------------------------------------------------------------------
New EXPORT function
------------------
* A new function, EXPORT, allows writing variables
and functions on other channels, the complement
of the IMPORT function.
app_amd
------------------
* An audio file to play during AMD processing can
now be specified to the AMD application or configured
in the amd.conf configuration file.
app_bridgewait
------------------
* Adds the n option to not answer the channel when
the BridgeWait application is called.
features
------------------
* The Bridge application now has the n "no answer" option
that can be used to prevent the channel from being
automatically answered prior to bridging.
func_strings
------------------
* Three new functions, TRIM, LTRIM, and RTRIM, are
now available for trimming leading and trailing
whitespace.
res_pjsip
------------------
* A new option named "peer_supported" has been added to the endpoint option
100rel. When set to this option, Asterisk sends provisional responses
reliably if the peer supports it. If the peer does not support reliable
provisional responses, Asterisk sends them normally.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
------------------------------------------------------------------------------
Transfer feature
------------------
* The following capabilities have been added to the
transfer feature:
- The transfer initiation announcement prompt can
now be customized in features.conf.
- The TRANSFER_EXTEN variable now can be set on the
transferer's channel in order to allow the transfer
function to automatically attempt to go to the extension
contained in this variable, if it exists. The transfer
context behavior is not changed (TRANSFER_CONTEXT is used
if it exists; otherwise the default context is used).
app_confbridge
------------------
* Adds the end_marked_any option which can be used
to kick users from a conference after any
marked user leaves (including marked users).
db
------------------
* The DBPrefixGet AMI action now allows retrieving
all of the DB keys beginning with a particular
prefix.
locks
------------------
* A new AMI event, DeadlockStart, is now available
when Asterisk is compiled with DETECT_DEADLOCKS,
and can indicate that a deadlock has occured.
res_geolocation
------------------
* * Added processing for the 'confidence' element.
* Added documentation to some APIs.
* removed a lot of complex code related to the very-off-nominal
case of needing to process multiple location info sources.
* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
one eprofile instead of a datastore of multiples.
* Plugged a huge leak in XML processing that arose from
insufficient documentation by the libxml/libxslt authors.
* Refactored stylesheets to be more efficient.
* Renamed 'profile_action' to 'profile_precedence' to better
reflect it's purpose.
* Added the config option for 'allow_routing_use' which
sets the value of the 'Geolocation-Routing' header.
* Removed the GeolocProfileCreate and GeolocProfileDelete
dialplan apps.
* Changed the GEOLOC_PROFILE dialplan function as follows:
* Removed the 'profile' argument.
* Automatically create a profile if it doesn't exist.
* Delete a profile if 'inheritable' is set to no.
* Fixed various bugs and leaks
* Updated Asterisk WiKi documentation.
Added 4 built-in profiles:
"<prefer_config>"
"<discard_config>"
"<prefer_incoming>"
"<discard_incoming>"
The profiles are empty except for having their precedence
set.
Added profile parameter "suppress_empty_ca_elements" that
will cause Civic Address elements that are empty to be
suppressed from the outgoing PIDF-LO document.
You can now specify the location object's format, location_info,
method, location_source and confidence parameters directly on
a profile object for simple scenarios where the location
information isn't common with any other profiles. This is
mutually exclusive with setting location_reference on the
profile.
Added an 'a' option to the GEOLOC_PROFILE function to allow
variable lists like location_info_refinement to be appended
to instead of replacing the entire list.
Added an 'r' option to the GEOLOC_PROFILE function to resolve all
variables before a read operation and after a Set operation.
res_musiconhold_answeredonly
------------------
* This change adds an option, answeredonly, that will prevent music
on hold on channels that are not answered.
res_pjsip
------------------
* TLS transports in res_pjsip can now reload their TLS certificate
and private key files, provided the filename of them has not
changed.
Applications
------------------
* added support for Danish syntax, playing the correct plural sound file

106191
ChangeLogs/historical/ChangeLog Normal file

File diff suppressed because it is too large Load Diff

View File

@ -45,7 +45,7 @@ redistribution of Asterisk source code obtained from Digium, you
should contact our licensing department to determine the necessary
steps you must take. For more information on this policy, please read:
https://www.sangoma.com/wp-content/uploads/Sangoma-Trademark-Policy.pdf
https://www.sangoma.com/wp-content/uploads/Sangoma-Trademark-Policy-1.pdf
If you have any questions regarding our licensing policy, please
contact us:

View File

@ -377,7 +377,7 @@ $(MOD_SUBDIRS_MENUSELECT_TREE):
+@$(SUBMAKE) -C $(@:-menuselect-tree=) SUBDIR=$(@:-menuselect-tree=) moduleinfo
+@$(SUBMAKE) -C $(@:-menuselect-tree=) SUBDIR=$(@:-menuselect-tree=) makeopts
$(SUBDIRS): makeopts .lastclean main/version.c include/asterisk/build.h include/asterisk/buildopts.h defaults.h
$(SUBDIRS): makeopts .lastclean main/version.c include/asterisk/build.h defaults.h
ifeq ($(findstring $(OSARCH), mingw32 cygwin ),)
main: third-party
@ -403,7 +403,7 @@ defaults.h: makeopts .lastclean build_tools/make_defaults_h
@cmp -s $@.tmp $@ || mv $@.tmp $@
@rm -f $@.tmp
main/version.c: FORCE menuselect.makeopts .lastclean
main/version.c: FORCE include/asterisk/buildopts.h menuselect.makeopts .lastclean
@build_tools/make_version_c > $@.tmp
@cmp -s $@.tmp $@ || mv $@.tmp $@
@rm -f $@.tmp
@ -545,7 +545,7 @@ INSTALLDIRS="$(ASTLIBDIR)" "$(ASTMODDIR)" "$(ASTSBINDIR)" "$(ASTCACHEDIR)" "$(AS
"$(ASTDATADIR)/firmware/iax" "$(ASTDATADIR)/images" "$(ASTDATADIR)/keys" \
"$(ASTDATADIR)/phoneprov" "$(ASTDATADIR)/rest-api" "$(ASTDATADIR)/static-http" \
"$(ASTDATADIR)/sounds" "$(ASTDATADIR)/moh" "$(ASTMANDIR)/man8" "$(AGI_DIR)" "$(ASTDBDIR)" \
"$(ASTDATADIR)/third-party" "${ASTDATADIR}/keys/stir_shaken"
"$(ASTDATADIR)/third-party" "${ASTDATADIR}/keys/stir_shaken" "${ASTDATADIR}/keys/stir_shaken/cache"
installdirs:
@for i in $(INSTALLDIRS); do \
@ -561,9 +561,9 @@ bininstall: _all installdirs $(SUBDIRS_INSTALL) main-bininstall
$(INSTALL) -m 755 contrib/scripts/astversion "$(DESTDIR)$(ASTSBINDIR)/"
$(INSTALL) -m 755 contrib/scripts/astgenkey "$(DESTDIR)$(ASTSBINDIR)/"
$(INSTALL) -m 755 contrib/scripts/autosupport "$(DESTDIR)$(ASTSBINDIR)/"
if [ ! -f /sbin/launchd ]; then \
./build_tools/install_subst contrib/scripts/safe_asterisk "$(DESTDIR)$(ASTSBINDIR)/safe_asterisk"; \
fi
ifneq ($(HAVE_SBIN_LAUNCHD),1)
./build_tools/install_subst contrib/scripts/safe_asterisk "$(DESTDIR)$(ASTSBINDIR)/safe_asterisk";
endif
ifneq ($(DISABLE_XMLDOC),yes)
$(INSTALL) -m 644 doc/core-*.xml "$(DESTDIR)$(ASTDATADIR)/documentation"
@ -1119,7 +1119,8 @@ ifeq ($(PYTHON),:)
else
@$(INSTALL) -d doc/rest-api
$(PYTHON) rest-api-templates/make_ari_stubs.py \
rest-api/resources.json .
--resources rest-api/resources.json --source-dir $(ASTTOPDIR) \
--dest-dir $(ASTTOPDIR)/doc/rest-api --docs-prefix ../
endif
check-alembic: makeopts

View File

@ -213,10 +213,10 @@ endif
# extern const size_t _binary_abc_def_xml_size;
%.o: %.xml
$(ECHO_PREFIX) echo " [LD] $^ -> $@"
$(CMD_PREFIX) $(CC) -g -nostartfiles -nodefaultlibs -nostdlib -r -Wl,-b,binary -o $@ $^
$(CMD_PREFIX) $(CC) -g -Wl,-znoexecstack -nostartfiles -nodefaultlibs -nostdlib -r -Wl,-b,binary -o $@ $^
%.o: %.xslt
$(ECHO_PREFIX) echo " [LD] $^ -> $@"
$(CMD_PREFIX) $(CC) -g -nostartfiles -nodefaultlibs -nostdlib -r -Wl,-b,binary -o $@ $^
$(CMD_PREFIX) $(CC) -g -Wl,-znoexecstack -nostartfiles -nodefaultlibs -nostdlib -r -Wl,-b,binary -o $@ $^
dist-clean:: clean

View File

@ -377,9 +377,8 @@ is set to no.
In Asterisk 12 and later, live_dangerously defaults to no.
[voip-security-webinar]: https://www.asterisk.org/security/webinar/
[blog-sip-security]: http://blogs.digium.com/2009/03/28/sip-security/
[voip-security-webinar]: https://docs.asterisk.org/Deployment/Important-Security-Considerations/Asterisk-Security-Webinars/
[blog-sip-security]: https://web.archive.org/web/20171030134647/http://blogs.digium.com/2009/03/28/sip-security/
[Strong Password Generator]: https://www.strongpasswordgenerator.com
[Filtering Data]: #filtering-data
[Proper Device Naming]: #proper-device-naming
@ -387,4 +386,4 @@ In Asterisk 12 and later, live_dangerously defaults to no.
[Reducing Pattern Match Typos]: #reducing-pattern-match-typos
[Manager Class Authorizations]: #manager-class-authorizations
[Avoid Privilege Escalations]: #avoid-privilege-escalations
[Important Security Considerations]: https://wiki.asterisk.org/wiki/display/AST/Important+Security+Considerations
[Important Security Considerations]: https://docs.asterisk.org/Deployment/Important-Security-Considerations/

View File

@ -20,7 +20,7 @@ more telephony interfaces than just Internet telephony. Asterisk also has a
vast amount of support for traditional PSTN telephony, as well.
For more information on the project itself, please visit the Asterisk
[home page] and the official [wiki]. In addition you'll find lots
[home page] and the official [documentation]. In addition you'll find lots
of information compiled by the Asterisk community at [voip-info.org].
There is a book on Asterisk published by O'Reilly under the Creative Commons
@ -48,7 +48,7 @@ ANY special hardware, not even a sound card) to install and run Asterisk.
Supported telephony hardware includes:
* All Analog and Digital Interface cards from [Sangoma]
* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
* QuickNet Internet PhoneJack and LineJack
* any full duplex sound card supported by ALSA, OSS, or PortAudio
* any ISDN card supported by mISDN on Linux
* The Xorcom Astribank channel bank
@ -258,7 +258,7 @@ Asterisk is a trademark of Sangoma Technologies Corporation
[home page]: https://www.asterisk.org
[support]: https://www.asterisk.org/support
[wiki]: https://wiki.asterisk.org/
[documentation]: https://docs.asterisk.org/
[mailing list]: http://lists.digium.com/mailman/listinfo/asterisk-users
[chan_dahdi.conf]: configs/samples/chan_dahdi.conf.sample
[voip-info.org]: http://www.voip-info.org/wiki-Asterisk
@ -269,4 +269,4 @@ Asterisk is a trademark of Sangoma Technologies Corporation
[CHANGES]: CHANGES
[configs]: configs
[doc]: doc
[Important Security Considerations]: https://wiki.asterisk.org/wiki/display/AST/Important+Security+Considerations
[Important Security Considerations]: https://docs.asterisk.org/Deployment/Important-Security-Considerations/

9
SECURITY.md Normal file
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@ -0,0 +1,9 @@
# Security Policy
## Supported Versions
The Asterisk project maintains a [documentation page](https://docs.asterisk.org/About-the-Project/Asterisk-Versions/) of releases. Each version is listed with its release date, security fix only date, and end of life date. Consult this wiki page to see if the version of Asterisk you are reporting a security vulnerability against is still supported.
## Reporting a Vulnerability
To report a vulnerability use the "Report a vulnerability" button under the "Security" tab of this project.

File diff suppressed because it is too large Load Diff

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@ -192,11 +192,13 @@ EXTERN int ooQ931Decode
screening indicators ;-) */
if(ie->discriminator == Q931CallingPartyNumberIE)
{
int numoffset=1;
OOTRACEDBGB1(" CallingPartyNumber IE = {\n");
if(ie->length < OO_MAX_NUMBER_LENGTH)
if(!(0x80 & ie->data[0])) numoffset = 2;
if( (ie->length >= numoffset) &&
(ie->length < OO_MAX_NUMBER_LENGTH) )
{
int numoffset=1;
if(!(0x80 & ie->data[0])) numoffset = 2;
memcpy(number, ie->data+numoffset,ie->length-numoffset);
number[ie->length-numoffset]='\0';
OOTRACEDBGB2(" %s\n", number);
@ -204,7 +206,7 @@ EXTERN int ooQ931Decode
ooCallSetCallingPartyNumber(call, number);
}
else{
OOTRACEERR3("Error:Calling party number too long. (%s, %s)\n",
OOTRACEERR3("Error:Calling party number outside range. (%s, %s)\n",
call->callType, call->callToken);
}
OOTRACEDBGB1(" }\n");
@ -214,7 +216,8 @@ EXTERN int ooQ931Decode
if(ie->discriminator == Q931CalledPartyNumberIE)
{
OOTRACEDBGB1(" CalledPartyNumber IE = {\n");
if(ie->length < OO_MAX_NUMBER_LENGTH)
if( (ie->length >= 1) &&
(ie->length < OO_MAX_NUMBER_LENGTH) )
{
memcpy(number, ie->data+1,ie->length-1);
number[ie->length-1]='\0';
@ -223,7 +226,7 @@ EXTERN int ooQ931Decode
ooCallSetCalledPartyNumber(call, number);
}
else{
OOTRACEERR3("Error:Calling party number too long. (%s, %s)\n",
OOTRACEERR3("Error:Calling party number outside range. (%s, %s)\n",
call->callType, call->callToken);
}
OOTRACEDBGB1(" }\n");

1
ami_test_client/.gitignore vendored Normal file
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@ -0,0 +1 @@
ami_test_client

339
ami_test_client/LICENSE Normal file
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@ -0,0 +1,339 @@
GNU GENERAL PUBLIC LICENSE
Version 2, June 1991
Copyright (C) 1989, 1991 Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
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The licenses for most software are designed to take away your
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This General Public License does not permit incorporating your program into
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20
ami_test_client/Makefile Normal file
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@ -0,0 +1,20 @@
CC = gcc
CFLAGS = -Wall -ggdb -Ilibc-jss
LDFLAGS = -lev -lm -lcrypto
OBJ=ami.o originate.o libc-jss/netsocket.o libc-jss/logger.o libc-jss/misc.o milenage.o
PROGS=ami_test_client
.PHONY: all
all: $(patsubst %, %.o, $(PROGS)) $(OBJ) $(PROGS)
%: %.o $(OBJ)
gcc $(CFLAGS) -o $@ $< $(OBJ) $(LDFLAGS)
clean:
rm -f *.o $(OBJ) $(PROGS)
volcon.o: volcon.c volcon_config.h
install: compile
install -s volcon /usr/local/bin

69
ami_test_client/NOTES.txt Normal file
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@ -0,0 +1,69 @@
BUGOK:
- Asterisk 1.4 esetén az originate.c rossz Hangup-cause kódot ad vissza, ha Zap
csatornán indítjuk a hívást. Rendre 19-et ad vissza. Nem az originate.c
hibája, mert az AMI dumpban is 19 látszódik. Local csatornán rendben működik.
TODO:
-első körben tesztelni kéne az ami.c-t a test_ami.c -vel. Írni valami kis célprogramot! pl. mellék állapot listázás
-ami.c-ben ami_event_list_t átnevezése valami kultúráltabb típusnévre, mert ez az API-ban megjelenik
-időzítőt betenni, hogy ha megszakad a kapcsolat az Asteriskkel, akkor ping timeout legyen
-időzítőt betenni, ami törlődik a sikeres authentikációnál
-Asterisk Call Manager/1.1 fejléc megérkezése előtt senki nem írhat az AMI socketre, mert ez lesz:
Asterisk Call Manager/1.1
Response: Error
ActionID: 1
Message: Missing action in request
-CLI response és az olyan több csomagos válaszok, mint pl. a DongleShowDevices kezelése
-ami_action-t ne lehessen regisztrálni, amíg nincs connected állapot
(vigyázni a beépített authentication ami_action-re!)
-ami_connect legyen hatástlan, ha már épp kapcsolódik vagy kapcsolódva van
-ami_event_unregister megírása
-ami dump events: response kulonvalasztasa
-event es response kulonvalasztasa keresesnel
-megvizsgálni, hogy az originate.c egyszerűbben működhet -e a linkedid változóval?
-Asterisk 1.4-et dobni, mert nem kell!
CLI formátum:
action: command
command: sip show domains
Response: Follows
Privilege: Command
SIP Domain support not enabled.
--END COMMAND--
SMS küldése
action: DongleSendPDU
ActionID: 59
Device: dongle0
PDU: 0031000B816002660304F000000010C8329BFD061DA74DD0F52D679343
Válasz a várakozósorból
Response: Success
ActionID: 59
Message: [dongle0] SMS queued for send
ID: 0x5e13c0
Elküldve
Event: DongleSMSStatus
Privilege: call,all
Device: dongle0
ID: 0x5e13c0
Status: Sent
Nincs elküldve
Event: DongleSMSStatus
Privilege: call,all
Device: dongle0
ID: 0x5c0888
Status: NotSent
JEGYZETEK
=========
A linkedid változó, pl: ${CHANNEL(LinkedID)} hívásonként és nem csatornánként
változik. Tehát a tovább kapcsolt, illetve Local channel által továbbvitt
hívások ugyan azt a LinkedID-t kapják.

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@ -0,0 +1,76 @@
# Asterisk Manager Interface client C library
libamievent is an asynchronous event-driven client library for Asterisk Manager
Interface written in C. It uses [libev](http://software.schmorp.de/pkg/libev.html) as event
loop backend.
With the libamievent you can send AMI commands and you can subscribe for
response to the command. When it arrives, the libamievent call the callback
function, which specified at subscription. The callback function allows you to
query AMI variables.
libamievent support AMI events. You can also specify a callback function of
what the libamievent is called when events are received.
Huge advantage of the libamievent, that the AMI commands and event names are
not hardcoded in the library. All commands, event names, parameters and
variables must be defined by printf-style strings with variable substitution.
## Requirements
* libev
For Debian users:
apt-get install libev-dev
For Gentoo users:
emerge -av libev
## Install
Currently, the Makefile is not prepared to carry out normal library. But, you
can build the example codes.
Clone the libamievent repo with all submodules.
git clone --recursive git://github.com/andrewjsi/libamievent
Compile the source, test and install:
make
## Using library
Sorry, the documentation still needs work, but in the meantime check out the sample
programs.
## Todo, Future
* more stability
* more examples
* build static and dynamic library
* well-written documentation
* the ability to be integrated with other systems as easy as possible
## Source code
The source code is well written as far as possible. We do not use tabs, instead
of 4 spaces is indented. All identifiers, including variables, function names
and macros written in English, but some comments and commits in Hungarian is,
because we are speaking and thinking in Hungarian. Nevertheless, we try to
write everything in English in the future.
## Contribution
It is an open source project, which is to be better and better. If you have any
ideas or find an error, or get stuck, you can contact us, please file a bug
report or send a pull-request!
## License
[_GPL2_](https://www.gnu.org/licenses/gpl-2.0.html)
(C) Copyright 2010-2014 Andras Jeszenszky, [JSS & Hayer
IT](http://www.jsshayer.hu) All Rights Reserved.

864
ami_test_client/ami.c Normal file
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/* ami.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
// debug infók (kommentezd, ha nem kell)
// #define CON_DEBUG
// csomagok dumpolása stdout-ra (kommentezd, ha nem kell)
// #define AMI_DEBUG_PACKET_DUMP
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <errno.h>
#include <ev.h>
#include <stdio.h> // TODO: kell ez?
#include <stdarg.h>
#include <sys/time.h> // gettimeofday()
#include "ami.h"
#include "debug.h"
#include "utlist.h"
#include "misc.h"
#include "logger.h"
// event rögzítése
void put_event (ami_event_t *event) {
ami_event_t *event_copy = (ami_event_t*)malloc(sizeof(ami_event_t));
if (!event_copy) {
conft("can't allocate event: %s, event lost", strerror(errno));
return;
}
//~ event tartalmának másolása az új event_copy számára lefoglalt területre.
//~ Emiatt lesz thread-safe a várakozó sor és a callback hívás.
memcpy(event_copy, event, sizeof(ami_event_t));
// bedobjuk a listába az új eseményt
DL_APPEND(event->ami->event_head, event_copy);
//~ Esedékessé tesszük az azonnali (0 sec) időzítőt, aminek hatására az event
//~ loop a callback futtatások után azonnal meghívja az invoke_events()
//~ függvényt, ami meghivogatja a sorban álló eventekhez tartozó callback
//~ eljárásokat. Majd a multithread fejlesztésénél ezen a ponton az
//~ ev_timer_start() helyett az ev_async() függvénnyel kell jelezni a másik
//~ szálban futó event loop-nak, hogy dolog van.
ev_timer_start(event->ami->loop, &event->ami->need_event_processing);
}
/* Felépítjük az event->field string tömböt, amiben az Asterisk által
küldött "változó: érték" párokat mentjük el úgy, hogy "változó", "érték",
"változó", "érték", ... A tömböt az ami->inbuf mutatókkal való
feldarabolásával és NULL byteok elhelyezésével kapjuk.
A sorvégeket lezárhatja \r\n és \n is egyaránt.
A legutolsó sor végét nem kötelező lezárni.
Ha az ami->inbuf tartalma:
Response: Success
Message: Authentication accepted
Akkor az ami->field:
{"Respone","Success","Message","Authentication accepted"}
field string tömb
max_field_size maximum ennyi darab string rakható a field tömbbe
field_len annyi lesz az értéke, ahány elem bekerült a field tömbbe
data innen olvassuk az adatokat és ezt a buffert daraboljuk fel és zárjuk le NULL-al
data_size data mérete
*/
//~ TODO: Hibás a függvény működése, ha a **field tömbünk mérete kicsi és a
//~ feldarabolás során nem férnek el benne a tokenek. Nincs segfault meg
//~ memóriahiba, hanem csak annyi történik, hogy az utolsó változó-érték pár
//~ értéke megkapja sortörésekkel együtt a maradék buffert. Ezt úgy lehetne
//~ megoldani, hogy a függvény nem bal-jobb oldalt vizsgál, hanem egy for ciklus
//~ NULL-ra állítja a ": " és a "\r" és "\n" karaktereket a teljes data-ban, majd
//~ csak ezután következne a feldarabolás mutatókkal.
//~
//~ aug 21: A fent leirt hiba sz'tem nem hiba.
//~
//~ aug 29: mi lenne, ha az utolsó változó-érték pár nem kapná meg a teljes buffert?
//~ vagy eleve netsocket_disconnect és feltakarítás kellene ide?
//~
//~ A függvény nem kezeli azt az esetet, amikor az AMI változó-érték párnak nincs
//~ értéke és nincs a változó utáni kettőspont után szóköz, hanem egyből újsor
//~ karakter. Ilyen eset áll fenn az RTCPSent esemény ReportBlock változójában.
void tokenize_field (int *field, int max_field_size, int *field_len, char *data, int data_size) {
enum {
LEFT,
RIGHT,
} inexpr = LEFT;
int len = 0; // visszatéréskor ezt mentjük el a *field_len -be
field[len++] = 0; // első pozíció a data legeleje, tehát 0
int i;
// összes \r karakter nullázása
for (i = 0; i < data_size; i++)
if (data[i] == '\r')
data[i] = '\0';
for (i = 0; i < data_size && len < max_field_size; i++) {
if (inexpr == LEFT) { // ": " bal oldalán vagyunk, változó
if (data[i] == ':' && data[i+1] == ' ') {
data[i] = '\0';
data[i+1] = '\0';
i += 2;
field[len++] = i;
inexpr = RIGHT;
}
}
if (inexpr == RIGHT) { // ": " jobb oldalán vagyunk, érték
if (data[i] == '\n') {
data[i] = '\0';
i += 1;
field[len++] = i;
inexpr = LEFT;
}
}
}
if (inexpr == LEFT)
len--;
*field_len = len;
// AMI bal és jobb értékek dumpolása
#ifdef AMI_DEBUG_PACKET_DUMP
int z;
for (z = 0; z < len; z++)
printf("tokenize_field ### %d - (%s)\n", z, &data[field[z]]);
printf("\n");
#endif
}
static char *type2name (enum ami_event_type type) {
switch (type) {
case AMI_EVENT: return "EVENT" ; break;
case AMI_RESPONSE: return "RESPONSE" ; break;
case AMI_CLIRESPONSE: return "CLIRESPONSE" ; break;
case AMI_CONNECT: return "CONNECT" ; break;
case AMI_DISCONNECT: return "DISCONNECT" ; break;
default: return "UNKNOWN";
}
}
// belső esemény kiváltása (AMI_CONNECT, AMI_DISCONNECT, stb...)
static void generate_local_event (ami_t *ami, enum ami_event_type type, const char *fmt, ...) {
ami_event_t event_tmp; // ideiglenes event // TODO: ha működik, akkor bevezetni az ami->event_tmp helyett lent is
ami_event_t *event = &event_tmp;
bzero(event, sizeof(*event));
//~ char buf[AMI_BUFSIZ];
va_list ap;
va_start(ap, fmt);
vsnprintf(event->data, sizeof(event->data), fmt, ap);
va_end(ap);
event->data[AMI_BUFSIZ-1] = '\0'; // védelem // TODO: kell ez?
event->data_size = strlen(event->data);
//~ printf("~~~ %s ~~~\n", event->data);
tokenize_field(
event->field,
sizeof(event->field) / sizeof(char*) - 1,
&event->field_size,
event->data,
event->data_size
);
ami_event_list_t *el = NULL;
// végigmegyünk a regisztrált eseményeken
DL_FOREACH(ami->ami_event_list_head, el) {
if (el->type == type) {
event->callback = el->callback;
event->userdata = el->userdata;
event->regby_file = el->regby_file;
event->regby_line = el->regby_line;
event->regby_function = el->regby_function;
event->regby_cbname = el->regby_cbname;
event->regby_udname = el->regby_udname;
event->ami = ami;
event->type = el->type;
put_event(event);
}
}
}
static void parse_cliresponse (ami_t *ami, int actionid, char *buf, int size) {
printf("***** CLI RESPONSE START *****\n");
printf("***** ActionID = %d\n", actionid);
int i;
for (i = 0; i < size; i++) {
putchar(buf[i]);
}
printf("***** CLI RESPONSE END *****\n\n");
}
// bejövő Response es Event feldolgozása
static void parse_input (ami_t *ami, char *buf, int size) {
ami_event_t *event = &ami->event_tmp;
bzero(event, sizeof(*event));
memcpy(event->data, buf, size);
event->data_size = size;
tokenize_field(
event->field,
sizeof(event->field) / sizeof(char*) - 1,
&event->field_size,
event->data,
size
);
char *var_response = ami_getvar(event, "Response");
char *var_event = ami_getvar(event, "Event");
/* * * RESPONSE * * */
if (!strlen(var_event)) {
char *action_id_str = ami_getvar(event, "ActionID");
if (action_id_str == NULL) {
con_debug("Missing ActionID in Response!");
return;
}
event->action_id = atoi(action_id_str);
if (!strcmp(var_response, "Success")) {
event->success = 1;
} else if (!strcmp(var_response, "Error")) {
event->success = 0;
} else {
con_debug("Unknown Response value: %s", var_response);
return;
}
con_debug("RESPONSE - success = %d, action_id = %d", event->success, event->action_id);
ami_event_list_t *el = NULL;
ami_event_list_t *eltmp = NULL;
DL_FOREACH_SAFE(ami->ami_event_list_head, el, eltmp) {
if (el->type != AMI_RESPONSE) // csak az AMI_RESPONSE típusú eseményeket vizsgáljuk
continue;
// event->action_id - Asterisktől érkezett ActionID
// el->action_id - adatbázisban szereplő ActionID
if (event->action_id == el->action_id) {
event->callback = el->callback;
event->userdata = el->userdata;
event->regby_file = el->regby_file;
event->regby_line = el->regby_line;
event->regby_function = el->regby_function;
event->regby_cbname = el->regby_cbname;
event->regby_udname = el->regby_udname;
event->ami = ami;
event->type = AMI_RESPONSE;
put_event(event);
DL_DELETE(ami->ami_event_list_head, el);
free(el);
return;
}
}
con_debug("Received ActionID=%d, but %d not found in ami_event_list_head!", event->action_id, event->action_id);
/* * * EVENT * * */
} else {
//~ printf("##### PARSE_INPUT EVENT #####\n");
ami_event_list_t *el;
// végigmegyünk a regisztrált eseményeken
DL_FOREACH(ami->ami_event_list_head, el) {
if (el->type != AMI_EVENT) // csak az AMI_EVENT típusú eseményeket vizsgáljuk
continue;
// regisztrációban definiált változó=érték párok száma
int need_found = el->field_size / 2; // minden találatnál dekrementálva lesz
//~ printf(" REG need_found=%d allevents=%d by %s:%d\n", need_found, el->allevents, el->regby_file, el->regby_line);
if (need_found || el->allevents) { // ha van mit keresnünk
int n, i;
// végigmegyünk a regisztráció változó=érték párjain
for (n = 0; n < el->field_size; n += 2) {
//~ printf(" _reg_ %s=%s\n", &el->data[el->field[n]], &el->data[el->field[n+1]]);
// végigmegyünk a bejövő csomag változó=érték párjain
for (i = 0; i < event->field_size; i += 2) {
//~ printf(" _eve_ %s=%s\n", &event->data[event->field[i]], &event->data[event->field[i+1]]);
// ha egyezik a regisztrált változó neve a csomag változó nevével
if (!strcmp(&el->data[el->field[n]], &event->data[event->field[i]])) {
// ha egyezik a regisztrált változó értéke a csomag változó értékével
if (!strcmp(&el->data[el->field[n+1]], &event->data[event->field[i+1]])) {
//~ printf(" !found\n");
need_found--;
}
}
}
}
//~ printf(" FIN need_found=%d\n", need_found);
// ha minden változó megtalálható volt és mindegyik értéke egyezett
// vagy "*" volt megadva a regisztrációnál (allevents)
if (need_found == 0 || el->allevents) {
event->callback = el->callback;
event->userdata = el->userdata;
event->regby_file = el->regby_file;
event->regby_line = el->regby_line;
event->regby_function = el->regby_function;
event->regby_cbname = el->regby_cbname;
event->regby_udname = el->regby_udname;
event->type = AMI_EVENT;
event->ami = ami;
put_event(event);
}
}
}
}
}
void ami_disconnect (ami_t *ami, const char *fmt, ...) {
char reason_text[AMI_BUFSIZ];
va_list ap;
va_start(ap, fmt);
vsnprintf(reason_text, sizeof(reason_text), fmt, ap);
va_end(ap);
if (netsocket_is_connected(ami->netsocket))
netsocket_disconnect_withevent(ami->netsocket, reason_text);
}
static void response_login (ami_event_t *response) {
ami_t *ami = response->ami;
con_debug("auth reply: success=%d %s (by %s() %s:%d)",
response->success,
ami_getvar(response, "Message"),
response->regby_function,
response->regby_file,
response->regby_line
);
if (!response->ami->authenticated) {
if (response->success) { // AUTH accepted
response->ami->authenticated = 1;
// TODO: itt kell a connect timeout időzítőt törölni
generate_local_event(ami,
AMI_CONNECT,
"Host: %s\nIP: %s\nPort: %d",
ami->host,
ami->netsocket->ip,
ami->port);
} else { // AUTH failed
netsocket_disconnect_withevent(response->ami->netsocket, "Authentication failed");
}
}
}
static void process_input (ami_t *ami) {
#ifdef AMI_DEBUG_PACKET_DUMP
int pdi;
printf("----- NETSOCKET INBUF START -----\n");
for (pdi = 0; pdi < ami->netsocket->inbuf_len; pdi++)
putchar(ami->netsocket->inbuf[pdi]);
printf("----- NETSOCKET INBUF END -----\n");
#endif
/*
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20
E v e n t : D i a l \r \n \r \n A c t i o n
\r \n \r \n
"Response: Follows\r\nPrivilege: Command\r\nActionID: %d\r\n"
"--END COMMAND--\r\n\r\n"
*/
// byte-onként végigmegyünk a netsocket bufferen
int i;
for (i = 0; i < ami->netsocket->inbuf_len; i++) {
// buffer overflow védelem
if (ami->inbuf_pos >= sizeof(ami->inbuf) - 1) {
printf("ELFOGYOTT A BUFFER!!!\n"); // TODO: netsocket_disconnect és feltakarítás
break;
}
// byte másolása netsocketből ami->inbuf -ba
ami->inbuf[ami->inbuf_pos] = ami->netsocket->inbuf[i];
/* AMI header vizsgalata. Biztonsagi okokbol ha mar authentikalt
allapotban vagyunk, akkor ezt a vizsgalatot kihagyjuk */
// TODO. lekezelni azt az esetet, amikor kezdésnek nem ezt a fejlécet kapjuk!
if (!ami->authenticated &&
(!strcmp(ami->inbuf, "Asterisk Call Manager/1.1\r\n") ||
!strcmp(ami->inbuf, "Asterisk Call Manager/1.0\r\n") ||
!strcmp(ami->inbuf, "Asterisk Call Manager/1.2\r\n") ||
!strcmp(ami->inbuf, "Asterisk Call Manager/1.3\r\n")))
{
bzero(ami->inbuf, sizeof(ami->inbuf));
ami->inbuf_pos = 0;
con_debug("Received \"Asterisk Call Manager\" header, sending auth...");
ami_action(ami, response_login, NULL,
"Action: Login\nUsername: %s\nSecret: %s\n",
ami->username, ami->secret);
bzero(ami->inbuf, sizeof(ami->inbuf));
ami->inbuf_pos = 0;
return;
}
// ha épp egy "Response: Follows" belsejében vagyunk
if (ami->cli_actionid > 0) {
// keressük a csomag végét
#define Q "--END COMMAND--\r\n\r\n"
#define QSIZE 19
if (ami->inbuf_pos >= QSIZE - 1) {
if (!strncmp(Q, &ami->inbuf[ami->inbuf_pos - (QSIZE - 1)], QSIZE)) {
// megtaláltuk, mehet a feldolgozóba
parse_cliresponse(ami, ami->cli_actionid, ami->inbuf, ami->inbuf_pos - 18);
bzero(ami->inbuf, sizeof(ami->inbuf));
ami->inbuf_pos = 0;
ami->cli_actionid = 0;
continue;
}
}
#undef Q
#undef QSIZE
// folytatjuk tovább a beolvasást
ami->inbuf_pos++;
continue;
}
/* Egy "Action: Command" csomagra (ami_cli() okozza) egy "Response:
Follows" válaszcsomag érkezik. Az ilyen csomagoknak speciális
formátuma van, ezért ezeket teljesen külön kell kezelni. Ezek a
csomagok a parse_input() helyett a parse_cliresponse() függvénynek
kerülnek át feldolgozásra. Az alábbi sscanf() megoldás megvizsgálja,
hogy az éppen beolvasás alatt álló csomag ilyen speciális "Response:
Follows" csomag lesz -e, illetve kiszedi belőle az ActionID-t. A
sscanf() az ami->inbuf baloldali illeszkedését vizsgálja és ha
tudja, akkor az ami->cli_actionid változóba menti el a kapott
ActionID-t. */
if (sscanf(ami->inbuf, "Response: Follows\r\nPrivilege: Command\r\nActionID: %d\r\n", &ami->cli_actionid) > 0) {
// az ami->inbuf jobboldali illeszkedését az strncmp()-vel vizsgáljuk
#define Q "\r\n"
#define QSIZE 2 // Q mérete
if (ami->inbuf_pos >= QSIZE - 1) {
// ha illeszkedik jobbról a \r\n, akkor tovább olvassuk az adatokat
if (!strncmp(Q, &ami->inbuf[ami->inbuf_pos - (QSIZE - 1)], QSIZE)) {
bzero(ami->inbuf, sizeof(ami->inbuf));
ami->inbuf_pos = 0;
continue; // ezen a ponton az ami->cli_actionid -ben ott figyel az ActionID
}
}
#undef Q
#undef QSIZE
/* Ide akkor kerülünk, ha a sscanf() (bal oldal) illeszkedett, de az
strncmp() (jobb oldal) nem. Ebben az esetben elképzelhető, hogy
a sscanf() hibásan kiolvassa az ActionID egy töredékét, ezért
biztos ami biztos, kinullázzuk. ezen a ponton lehetséges, hogy a
sscanf() az ActinID csak egy töredékét szedte ki, ezért nullázunk. */
ami->cli_actionid = 0;
// folytatjuk tovább a beolvasást
ami->inbuf_pos++;
continue;
}
/* Ha van elég adat, hogy az ami->inbuf -ban 3 byte-ot visszaléphessünk,
akkor megvizsgáljuk, hogy vajon éppen egy csomag lezárásánál állunk
-e, azaz az ami->inbuf legutolsó 4 byte-ja megegyezik ezzel:
"\r\n\r\n". Ha igen, akkor az azt jelenti, hogy az ami->inbuf pont
egy teljes csomagot tartalmaz, amit elküldünk a parse_input()-nak,
majd kinullázzuk a teljes ami->inbuf buffert és az ami->inbuf_pos
pozicionáló változót. Ezután folytatjuk a következő csomag byte-
onkénti olvasását. Ha már nincs a netsocket->inbuf -ban feldolgozandó
cucc, akkor a for ciklus kilép és majd a következő körben
folytatódik az olvasás. */
/* TODO: Logikailag nem korrekt ez a megoldás! Segfault veszély
ami_event_unregister() után. A process_input() és parse_input()
páros a teljes netsocket->inbuf -ban lévő cuccot egyetlen egy körben
feldolgozza. Ebben egyszerre több csomag is lehet. Az érdekes
eseményeket a put_event() a saját listájába tolja és csak a
következő körben lesz callback hívás. Tegyük fel, hogy egyszerre 2
csomag érkezik. Az első csomagban lévő esemény callback függvénye
megrendel egy új eseményt, amire történetesen pont a második csomag
illeszkedne. De mivel a megrendelés előtt már megtörtént az
összehasonlítás, szűrés és a futtatandó események kiválasztása,
ezért erről a második eseményről le fog maradni a hívó. Másképpen
szólva egy-egy megrendelésnek (vagy lemondásnak) csak a teljes
put_event() által karbantartott lista (callback lista) lefuttatása
után lesz hatása. Ez eseményről való lemaradást okozhat, illetve
lemondásnál segfaultot is, ugyanis ha történik egy
ami_event_unregister() akkor ezután még a put_event() által
karbantartott listából lefuthat a (már lemondott) callback. Ötlet:
valami olyan megoldás kéne, hogy még itt a process_input /
parse_input szintjén ha fennakad a szűrőn egy esemény, akkor a
put_event() regisztráció után álljon le a parse_input() és az event
loop hívja meg a need_event_processing-et. És majd csak ezután
folytatódjon a parse_input() vizsgálódása. Vaaagy... egy merészebb
ötlet. A bejövő AMI buffer visszamenőleg addig legyen eltárolva,
amíg az invoke_events még foglalkozik a callback hívásokkal.
Multithread környezetben az invoke_events a megrendelő szálában fog
futni. Elképzelhető, hogy az invoke_events-ből kellene vizsgálni
azt, hogy az éppen bejövő AMI eseményt kell -e futtatni. */
#define Q "\r\n\r\n"
#define QSIZE 4
if (ami->inbuf_pos >= QSIZE - 1) {
if (!strncmp(Q, &ami->inbuf[ami->inbuf_pos - (QSIZE - 1)], QSIZE)) {
parse_input(ami, ami->inbuf, ami->inbuf_pos - 1); // -1 azért, hogy ne menjen át a legutolsó \r\n-ből az \r (érthető?)
bzero(ami->inbuf, sizeof(ami->inbuf));
ami->inbuf_pos = 0;
continue; // ne jusson el az ami->inbuf_pos++ -ig :)
}
}
#undef Q
#undef QSIZE
ami->inbuf_pos++;
}
}
static void netsocket_callback (netsocket_t *netsocket, int event) {
ami_t *ami = netsocket->userdata;
int was_authenticated = 0;
switch (event) {
case NETSOCKET_EVENT_CONNECT:
con_debug("Connected to %s (%s) port %d",
netsocket->host,
netsocket->ip,
netsocket->port
);
break;
case NETSOCKET_EVENT_DISCONNECT:
was_authenticated = ami->authenticated;
// TODO: itt kell alaphelyzetbe állítani az ami-t.
// disconnect esemény szétkűrtölése előtt
ami->authenticated = 0;
generate_local_event(ami,
AMI_DISCONNECT,
"Host: %s\nIP: %s\nPort: %d\nReason: %s\nWasAuthenticated: %d",
netsocket->host,
(netsocket->ip) ? netsocket->ip : "",
netsocket->port,
netsocket->disconnect_reason,
was_authenticated
);
if (netsocket->connected) {
con_debug("Disconnected from %s: %s",
netsocket->host,
netsocket->disconnect_reason
);
} else {
con_debug("Can't connect to %s[%s]:%d %s",
netsocket->host,
(netsocket->ip) ? netsocket->ip : "",
netsocket->port,
netsocket->disconnect_reason
);
}
break;
case NETSOCKET_EVENT_READ:
process_input(ami);
break;
}
}
// hívja az ami->need_event_processing azonnali időzítő
static void invoke_events (EV_P_ ev_io *w, int revents) {
ami_t *ami = w->data;
ami_event_t *event, *tmp;
DL_FOREACH_SAFE(ami->event_head, event, tmp) {
if (event->callback != NULL) {
con_debug("call %s()", event->regby_cbname);
event->callback(event);
con_debug("end %s()", event->regby_cbname);
}
DL_DELETE(ami->event_head, event);
free(event);
}
}
// 6 byte-os random hexa stringet masol az ami->uuid bufferbe
// TODO: egy rendes, unique ID-t visszaado fuggvenyt irni ehelyett a random vacak helyett
// pl. az util-linux-ng csomagban levo libuuid segitsegevel
static void generate_uuid (char *dst, size_t size) {
struct timeval tv;
int num;
char tmp[16];
gettimeofday(&tv, NULL);
srand(tv.tv_usec * tv.tv_sec);
num = rand();
snprintf(tmp, sizeof(tmp), "%x", num);
tmp[6] = '\0';
strncpy(dst, tmp, size);
}
void connect_delayed (EV_P_ ev_io *w, int revents) {
ami_t *ami = w->data;
con_debug("invoked connect by timer");
ev_timer_stop(ami->loop, &ami->t_connect_delayed);
ami_connect(ami);
}
// delay: millisec
void ami_connect_delayed (ami_t *ami, int delay) {
con_debug("connect after %d ms ...", delay);
ev_timer_stop(ami->loop, &ami->t_connect_delayed);
ev_timer_set(&ami->t_connect_delayed, (float)((float)delay / (float)1000), 0);
ev_timer_start(ami->loop, &ami->t_connect_delayed);
}
ami_t *ami_new (struct ev_loop *loop) {
ami_t *ami = malloc(sizeof(*ami));
if (ami == NULL) {
con_debug("ami_new() returned NULL");
return NULL;
}
bzero(ami, sizeof(*ami)); // minden NULL
// AMI UUID
generate_uuid(ami->uuid, sizeof(ami->uuid));
// ha meg van adva a loop paraméter, akkor azt használjuk eseménykezelőnek
// ellenkező esetben az alapértelmezett eseménykezelőt
ami->loop = (loop != NULL) ? loop : ev_default_loop(0);
// default értékek
strncpy(ami->host, AMI_DEFAULT_HOST, sizeof(ami->host) - 1);
ami->port = AMI_DEFAULT_PORT;
if (!(ami->netsocket = netsocket_new(netsocket_callback, ami, ami->loop))) {
con_debug("netsocket_new returned NULL");
}
netsocket_host(ami->netsocket, AMI_DEFAULT_HOST);
netsocket_port(ami->netsocket, AMI_DEFAULT_PORT);
ami->need_event_processing.data = ami; // ami objektum így kerül az invoke_events-be
ev_timer_init(&ami->need_event_processing, (void*)invoke_events, 0, 0);
ami->t_connect_delayed.data = ami;
ev_timer_init(&ami->t_connect_delayed, (void*)connect_delayed, 0, 0);
return ami;
}
void ami_destroy(ami_t *ami) {
netsocket_destroy(ami->netsocket);
}
void ami_credentials (ami_t *ami, const char *username, const char *secret, const char *host, const char *port) {
if (username != NULL)
strncpy(ami->username, username, sizeof(ami->username) - 1);
if (secret != NULL)
strncpy(ami->secret, secret, sizeof(ami->secret) - 1);
if (host != NULL)
strncpy(ami->host, host, sizeof(ami->host) - 1);
if (port != NULL) {
int port_tmp = atoi(port);
if (port_tmp > 0 || port_tmp < 65536)
ami->port = port_tmp;
}
}
void ami_connect (ami_t *ami) {
netsocket_host(ami->netsocket, ami->host);
netsocket_port(ami->netsocket, ami->port);
netsocket_connect(ami->netsocket);
}
int ami_printf (ami_t *ami, const char *fmt, ...) {
char buf[AMI_BUFSIZ];
va_list ap;
va_start(ap, fmt);
vsnprintf(buf, sizeof(buf), fmt, ap);
va_end(ap);
int field[AMI_FIELD_SIZE];
int field_size;
tokenize_field(
field,
sizeof(field) / sizeof(char*) - 1,
&field_size,
buf,
strlen(buf)
);
char packet[AMI_BUFSIZ];
int i;
strcpy(packet, "");
for (i = 0; i < field_size; i += 2)
concatf(packet, "%s: %s\r\n", &buf[field[i]], &buf[field[i+1]]);
concat(packet, "\r\n");
if (ami->netsocket != NULL) {
#ifdef AMI_DEBUG_PACKET_DUMP
printf("----- NETSOCKET WRITE START ------\n");
printf("%s", packet);
printf("----- NETSOCKET WRITE END ------\n");
#endif
return netsocket_printf(ami->netsocket, "%s", packet);
} else {
return -1;
}
}
ami_event_list_t *_ami_action (ami_t *ami, void *callback, void *userdata, char *file, int line, const char *function, const char *cbname, const char *udname, const char *fmt, ...) {
char buf[AMI_BUFSIZ];
va_list ap;
va_start(ap, fmt);
vsnprintf(buf, sizeof(buf), fmt, ap);
va_end(ap);
buf[AMI_BUFSIZ-1] = '\0'; // védelem
if (callback != NULL) {
ami_event_list_t *el = malloc(sizeof(ami_event_list_t));
bzero(el, sizeof(*el)); // NULL, NULL, NULL :)
el->callback = callback;
el->userdata = userdata;
el->type = AMI_RESPONSE;
el->regby_file = file;
el->regby_line = line;
el->regby_function = function;
el->regby_cbname = cbname;
el->regby_udname = udname;
ami->action_id++; // új ActionID
el->action_id = ami->action_id;
ami_printf(ami, "ActionID: %d\n%s", ami->action_id, buf);
con_debug("registered action #%d, callback: %s()", el->action_id, el->regby_cbname);
DL_APPEND(ami->ami_event_list_head, el);
return el;
} else {
ami_printf(ami, "%s", buf);
return NULL;
}
}
ami_event_list_t *_ami_event_register (ami_t *ami, void *callback, void *userdata, char *file, int line, const char *function, const char *cbname, const char *udname, const char *fmt, ...) {
ami_event_list_t *el = malloc(sizeof(ami_event_list_t));
bzero(el, sizeof(*el)); // NULL, NULL, NULL :)
va_list ap;
va_start(ap, fmt);
vsnprintf(el->data, sizeof(el->data), fmt, ap);
va_end(ap);
el->callback = callback;
el->userdata = userdata;
el->regby_file = file;
el->regby_line = line;
el->regby_function = function;
el->regby_cbname = cbname;
el->regby_udname = udname;
// belső esemény: Connect
if (!strcmp(el->data, "Connect")) {
el->type = AMI_CONNECT;
// belső esemény: Disconnect
} else if (!strcmp(el->data, "Disconnect")) {
el->type = AMI_DISCONNECT;
// Minden Asterisk esemény szűrés nélkül
} else if (!strcmp(el->data, "*")) {
el->type = AMI_EVENT;
el->allevents = 1;
// Asterisk esemény, feltételek feldarabolása
} else {
el->type = AMI_EVENT;
tokenize_field(
el->field,
sizeof(el->field) / sizeof(char*) - 1,
&el->field_size,
el->data,
sizeof(el->data)
);
}
DL_APPEND(ami->ami_event_list_head, el);
con_debug("EVENT registered, callback: %s by %s() in %s line %d",
el->regby_cbname, el->regby_function, el->regby_file, el->regby_line);
return el;
}
void ami_event_unregister(ami_t *ami, ami_event_list_t *el) {
if (el == NULL) {
con_debug("attempting to unregister NULL pointer event!");
return;
}
con_debug("EVENT unregistered, callback: %s()", el->regby_cbname);
DL_DELETE(ami->ami_event_list_head, el);
free(el);
}
void ami_event_dump (ami_event_t *event) {
printf(
"Incoming %s /0x%lx/\n"
" Registered by %s() in %s line %d\n"
" Callback: %s() /0x%lx/, Userdata: %s /0x%lx/\n"
" success=%d action_id=%d data_size=%d field_size=%d\n"
, type2name(event->type), (unsigned long)event
, event->regby_function, event->regby_file, event->regby_line
, event->regby_cbname, (unsigned long)event->callback, event->regby_udname, (unsigned long)event->userdata
, event->success, event->action_id, event->data_size, event->field_size
);
int i;
for (i = 0; i < event->field_size; i += 2)
printf(" %-16s %s\n", &event->data[event->field[i]], &event->data[event->field[i+1]]);
printf("\n");
}
void ami_dump_event_list_element (ami_event_list_t *el) {
printf(
"Registered %s /0x%lx/\n"
" Registered by %s() in %s line %d\n"
" Callback: %s() /0x%lx/, Userdata: %s /0x%lx/\n"
" action_id=%d field_size=%d\n"
, type2name(el->type), (unsigned long)el
, el->regby_function, el->regby_file, el->regby_line
, el->regby_cbname, (unsigned long)el->callback, el->regby_udname, (unsigned long)el->userdata
, el->action_id, el->field_size
);
int i;
for (i = 0; i < el->field_size; i += 2)
printf(" %-16s %s\n", &el->data[el->field[i]], &el->data[el->field[i+1]]);
printf("\n");
}
void ami_dump_lists (ami_t *ami) {
printf("** REGISTERED AMI EVENTS **\n");
ami_event_list_t *el;
DL_FOREACH(ami->ami_event_list_head, el)
ami_dump_event_list_element(el);
}
char *ami_getvar (ami_event_t *event, char *var) {
int i;
for (i = 0; i < event->field_size; i += 2) {
if (!strcmp(&event->data[event->field[i]], var)) {
if (&event->data[event->field[i+1]] != NULL) {
return &event->data[event->field[i+1]];
} else {
return "";
}
}
}
return "";
}
void ami_strncpy (ami_event_t *event, char *dest, char *var, size_t maxsize) {
}

135
ami_test_client/ami.h Normal file
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/* ami.h
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#ifndef AMI_H_INCLUDED
#define AMI_H_INCLUDED
#include "netsocket.h"
// TODO: doksiba AMI_DEBUG_PACKET_DUMP
#ifndef AMI_DEFAULT_HOST
#define AMI_DEFAULT_HOST "localhost"
#endif
#ifndef AMI_DEFAULT_PORT
#define AMI_DEFAULT_PORT 5038
#endif
#ifndef AMI_BUFSIZ
#define AMI_BUFSIZ 4096
#endif
#ifndef AMI_FIELD_SIZE
#define AMI_FIELD_SIZE 300
#endif
enum ami_event_type {
AMI_EVENT = 1,
AMI_RESPONSE,
AMI_CLIRESPONSE,
AMI_CONNECT,
AMI_DISCONNECT,
};
// ha változik, akkor egyeztess az ami.c ami_dump_lists() függvénnyel!
typedef struct ami_event_list_t {
struct ami_event_list_t *prev;
struct ami_event_list_t *next;
void (*callback)(void*);
void *userdata;
int field[AMI_FIELD_SIZE];
int field_size;
char data[AMI_BUFSIZ];
char *regby_file;
int regby_line;
const char *regby_function;
const char *regby_cbname;
const char *regby_udname;
unsigned int action_id;
enum ami_event_type type;
int allevents; // 1 lesz, ha a megrendelo szures nelkul az osszes AMI esemenyt keri
} ami_event_list_t;
typedef struct ami_event_t {
struct ami_event_t *prev;
struct ami_event_t *next;
struct ami_t *ami;
int success; // csak "Response: Success" esetén lesz egy, tehát biztos hogy volt Response és az értéke Success volt
int field[AMI_FIELD_SIZE];
int field_size;
char data[AMI_BUFSIZ];
int data_size;
void (*callback)(void*);
void *userdata;
unsigned int action_id;
char *regby_file;
int regby_line;
const char *regby_function;
const char *regby_cbname;
const char *regby_udname;
enum ami_event_type type;
} ami_event_t;
typedef struct ami_t {
char host[64]; // Asterisk host
int port; // Asterisk Manager Interface port
char username[32]; // AMI User
char secret[32]; // AMI Password
netsocket_t *netsocket; // Netsocket objektum
char disconnect_reason[64]; // ???
ami_event_list_t *ami_event_list_head; // megrendelt események
struct ev_loop *loop; // eseményhurok
ev_timer need_event_processing; // azonnali időzítő az események kiküldéséhez
ev_timer t_connect_delayed; // késleltetett connect időzítő
char inbuf[AMI_BUFSIZ]; // bejövő buffer
int inbuf_pos; // bejövő buffer pozíciója
struct ami_event_t *event_head; // esemény várakozósor
struct ami_event_t event_tmp; // itt készítünk új eseményt, amit aztán a várakozósorba másolunk
int authenticated; // 1 lesz sikeres login után
unsigned int action_id; // soron következő használható ActionID
char uuid[16]; // AMI sajat belso ID
int cli_actionid; // process_input() itt jegyzi meg a Response: Follows ActionID-t
} ami_t;
ami_t *ami_new (struct ev_loop *loop);
void ami_credentials (ami_t *ami, const char *username, const char *secret, const char *host, const char *port);
void ami_destroy(ami_t *ami);
void ami_connect (ami_t *ami);
void ami_connect_delayed (ami_t *ami, int delay);
int ami_printf (ami_t *ami, const char *fmt, ...);
#define ami_action(ami, callback, userdata, ...) \
_ami_action(ami, callback, userdata, __FILE__, __LINE__, __FUNCTION__, #callback, #userdata, __VA_ARGS__)
ami_event_list_t *_ami_action (ami_t *ami, void *callback, void *userdata, char *file, int line, const char *function, const char *cbname, const char *udname, const char *fmt, ...);
#define ami_event_register(ami, callback, userdata, ...) \
_ami_event_register(ami, callback, userdata, __FILE__, __LINE__, __FUNCTION__, #callback, #userdata, __VA_ARGS__)
ami_event_list_t *_ami_event_register (ami_t *ami, void *callback, void *userdata, char *file, int line, const char *function, const char *cbname, const char *udname, const char *fmt, ...);
void ami_event_unregister(ami_t *ami, ami_event_list_t *el);
char *ami_getvar (ami_event_t *event, char *var);
#define ami_strcpy(event,dest,var) ami_strncpy(event,dest,var,sizeof(dest))
void ami_strncpy (ami_event_t *event, char *dest, char *var, size_t maxsize);
void ami_dump_lists (ami_t *ami);
void ami_event_dump (ami_event_t *el);
void ami_disconnect (ami_t *ami, const char *fmt, ...);
#endif // #ifndef AMI_H_INCLUDED

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#include <stdio.h>
#include <string.h>
#include <ev.h>
#include <stdlib.h>
#include <errno.h>
#include "ami.h"
#include "debug.h"
#include "utlist.h"
#define CON_DEBUG
#include "logger.h"
#include "milenage.h"
ev_timer timer;
ev_timer timer_reconnect;
ami_t *ami;
int dstatus_n = 2;
//~ static void ami_event_callback (ami_event_t *ame) {
//~ char *userdata = (char*)ame->userdata;
//~ char *status = ami_getvar(ame, "Status");
//~
//~ char status2[64];
//~ strncpy(status2, ami_getvar(ame, "Status"), 64);
//~ ami_strncpy(ame, status2, "Status", 64);
//~
//~ ami_event_unregister(ame);
//~ }
void utproba () {
typedef struct st {
struct st *prev;
struct st *next;
char str[32];
} st;
st *head = NULL;
st *el;
st *x10, *x15;
int i;
for (i = 0; i < 20; i++) {
el = (st*)malloc(sizeof(*el));
bzero(el, sizeof(*el));
sprintf(el->str, "str %d", i);
if (i == 10)
x10 = el;
if (i == 15)
x15 = el;
DL_APPEND(head, el);
}
st *eltmp;
DL_FOREACH_SAFE(head, el, eltmp) {
if (el == x10) {
st *csere = (st*)malloc(sizeof(st));
strcpy(csere->str, "csere1");
DL_APPEND(head, csere);
csere = (st*)malloc(sizeof(st));
strcpy(csere->str, "csere2");
DL_APPEND(head, csere);
DL_DELETE(head, el);
free(el);
}
if (el == x15) {
DL_DELETE(head, el);
free(el);
}
}
DL_FOREACH_SAFE(head, el, eltmp) {
printf("s %s s\n", el->str);
DL_DELETE(head, el);
free(el);
}
return;
}
int do_register = 0;
void event_response (ami_event_t *event) {
const char *message_str = ami_getvar(event, "Message");
printf("response: %s\n", message_str);
}
void event_fullybooted (ami_event_t *event) {
printf("*** Fully booted ***\n");
if (do_register)
return;
do_register = 1;
ami_action(ami, event_response, NULL, "Action: PJSIPRegister\nRegistration: volte_ims");
ami_action(ami, event_response, NULL, "Action: PJSIPAccessNetworkInfo\nRegistration: volte_ims\nInfo: hello");
}
int hex_to_octet_string(const char *name, const char *input, uint8_t *output, size_t output_size)
{
int i, n;
if (!input || !input[0]) {
printf("Missing value for hex string '%s'.\n", name);
return -1;
}
i = n = 0;
while (*input) {
if (i == output_size) {
printf("Value for hex string '%s' too long, expecting %zu bytes.\n", name,
output_size);
return -1;
}
if (*input >= '0' && *input <= '9')
output[i] = (output[i] << 4) | (*input - '0');
else if (*input >= 'a' && *input <= 'f')
output[i] = (output[i] << 4) | (*input - 'a' + 10);
else if (*input >= 'A' && *input <= 'F')
output[i] = (output[i] << 4) | (*input - 'A' + 10);
else {
printf("Value for hex string '%s' has invalid character '%c'.\n", name, *input);
return -1;
}
if (++n == 2) {
n = 0;
i++;
}
input++;
}
if (i < output_size) {
printf("Value for hex string '%s' too short, expecting %zu bytes.\n", name, output_size);
return -1;
}
return 0;
}
void octest_string_to_hex(uint8_t *input, size_t input_size, char *output)
{
while(input_size) {
sprintf(output, "%02x", *input);
input++;
input_size--;
output += 2;
}
}
static const char *usim_opc = "775A1F887D2AD66F9719C2C79F847B50";
static const char *usim_k = "D534E07854B75E475C667A856AA31F9C";
static const char *usim_sqn = "000000011000";
void event_authrequest (ami_event_t *event) {
const char *rand_str, *autn_str;
uint8_t opc[16], k[16], sqn[6], rand[16], autn[16], res[8], ik[16], ck[16], auts[14];
char res_str[256], ik_str[256], ck_str[256], auts_str[256];
size_t res_len = 8;
rand_str = ami_getvar(event, "RAND");
autn_str = ami_getvar(event, "AUTN");
int rc;
if (!rand_str || !autn_str) {
printf("Missing parameters in AuthRequest\n");
return;
}
if (hex_to_octet_string("RAND", rand_str, rand, sizeof(rand))) {
printf("Invalid RAND parameter in AuthRequest\n");
return;
}
if (hex_to_octet_string("AUTN", autn_str, autn, sizeof(autn))) {
printf("Invalid AUTN parameter in AuthRequest\n");
return;
}
if (hex_to_octet_string("OPC", usim_opc, opc, sizeof(opc))) {
printf("Invalid OPC parameter in AuthRequest\n");
return;
}
if (hex_to_octet_string("K", usim_k, k, sizeof(k))) {
printf("Invalid K parameter in AuthRequest\n");
return;
}
if (hex_to_octet_string("SQN", usim_sqn, sqn, sizeof(sqn))) {
printf("Invalid SQN parameter in AuthRequest\n");
return;
}
rc = milenage_check(opc, k, sqn, rand, autn, ik, ck, res, &res_len, auts);
octest_string_to_hex(res, sizeof(res), res_str);
octest_string_to_hex(ik, sizeof(ik), ik_str);
octest_string_to_hex(ck, sizeof(ck), ck_str);
octest_string_to_hex(auts, sizeof(auts), auts_str);
if (rc == -2) {
printf("*** Sending AuthResponse with authentication token ***\n");
ami_action(ami, NULL, NULL, "Action: AuthResponse\nRegistration: volte_ims\nAUTS: %s", auts_str);
} else if (!rc) {
printf("*** Sending AuthResponse with authentication result and keys ***\n");
ami_action(ami, NULL, NULL, "Action: AuthResponse\nRegistration: volte_ims\nRES: %s\nCK: %s\nIK: %s", res_str, ck_str, ik_str);
} else {
printf("*** Sending AuthResponse with authentication error\n");
ami_action(ami, NULL, NULL, "Action: AuthResponse\nRegistration: volte_ims");
}
}
void reconnect_ami (ami_event_t *event) {
printf("*** Reconnecting... ***\n");
ev_timer_stop(EV_DEFAULT, &timer_reconnect);
ami_connect(ami);
}
void event_connect (ami_event_t *event) {
printf("*** CONNECTED ***\n");
printf(" semmi = %s\n host = %s\n ip = %s\n port = %s\n",
ami_getvar(event, "semmi"),
ami_getvar(event, "Host"),
ami_getvar(event, "IP"),
ami_getvar(event, "Port"));
}
void event_disconnect (ami_event_t *event) {
printf("*** DISCONNECTED ***\n");
printf(" reason = %s\n host = %s\n ip = %s\n port = %s\n",
ami_getvar(event, "Reason"),
ami_getvar(event, "Host"),
ami_getvar(event, "IP"),
ami_getvar(event, "Port"));
ev_timer_again(EV_DEFAULT, &timer_reconnect);
ev_timer_start(EV_DEFAULT, &timer_reconnect);
}
int main (int argc, char *argv[]) {
ami = ami_new(EV_DEFAULT);
if (ami == NULL) {
con_debug("ami_new() returned NULL");
return 1;
}
char host[128];
if (argc < 2) {
//~ printf("usage: %s <host>\n", argv[0]);
printf("Default host used!\n");
strcpy(host, "127.0.0.1");
} else {
strcpy(host, argv[1]);
}
printf("Connecting to %s ...\n", host);
ami_credentials(ami, "jolly", "svenja", host, "5038");
ami_connect(ami);
ami_event_register(ami, event_fullybooted, NULL, "Event: FullyBooted");
ev_timer_init(&timer_reconnect, (void*)reconnect_ami, 3, 3);
//~ ev_timer_start(EV_DEFAULT, &timer_reconnect);
ami_event_register(ami, event_connect, NULL, "Connect");
ami_event_register(ami, event_disconnect, NULL, "Disconnect");
ami_event_register(ami, event_authrequest, NULL, "Event: AuthRequest");
printf("\n");
ami_dump_lists(ami);
ami_action(ami, NULL, NULL, "Action: Login\nUsername: jolly\nSecret: svenja");
ev_loop(EV_DEFAULT, 0);
ami_destroy(ami);
return 0;
}

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2013-09-17 - netsocket
==========
A netsocket mostantól nem fogad el kívülről értékeket, tehát nem szabad
közvetlenül a programból manipulálni a netsocket_t objektumot. A host, lhost,
port, lport értékeket az alábbi függvények segítségével lehet beállítani. Ennek
oka az, hogy eddig a netsocket_t mutatókban tárolta a host és lhost stringeket,
ami azt okozta, hogy ha a főprogramban megváltoztattuk a hostot, akkor a
netsocket-ben is megváltozott, hisz ugyan arra mutattak. Mostantól a
netsocket_t saját char[] tömbben tárolja ezeket az értékeket, így nem lesz
hatással semmilyen külső változtatás, illetve nem fog összekeveredni.
Eddig így kellett hívni:
netsocket->host = host;
netsocket->port = port;
netsocket->lhost = lhost;
netsocket->lport = lport;
Mostantól pedig így kell:
netsocket_host(netsocket, host);
netsocket_port(netsocket, port);
netsocket_lhost(netsocket, lhost);
netsocket_lport(netsocket, lport);

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GNU GENERAL PUBLIC LICENSE
Version 2, June 1991
Copyright (C) 1989, 1991 Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
Preamble
The licenses for most software are designed to take away your
freedom to share and change it. By contrast, the GNU General Public
License is intended to guarantee your freedom to share and change free
software--to make sure the software is free for all its users. This
General Public License applies to most of the Free Software
Foundation's software and to any other program whose authors commit to
using it. (Some other Free Software Foundation software is covered by
the GNU Lesser General Public License instead.) You can apply it to
your programs, too.
When we speak of free software, we are referring to freedom, not
price. Our General Public Licenses are designed to make sure that you
have the freedom to distribute copies of free software (and charge for
this service if you wish), that you receive source code or can get it
if you want it, that you can change the software or use pieces of it
in new free programs; and that you know you can do these things.
To protect your rights, we need to make restrictions that forbid
anyone to deny you these rights or to ask you to surrender the rights.
These restrictions translate to certain responsibilities for you if you
distribute copies of the software, or if you modify it.
For example, if you distribute copies of such a program, whether
gratis or for a fee, you must give the recipients all the rights that
you have. You must make sure that they, too, receive or can get the
source code. And you must show them these terms so they know their
rights.
We protect your rights with two steps: (1) copyright the software, and
(2) offer you this license which gives you legal permission to copy,
distribute and/or modify the software.
Also, for each author's protection and ours, we want to make certain
that everyone understands that there is no warranty for this free
software. If the software is modified by someone else and passed on, we
want its recipients to know that what they have is not the original, so
that any problems introduced by others will not reflect on the original
authors' reputations.
Finally, any free program is threatened constantly by software
patents. We wish to avoid the danger that redistributors of a free
program will individually obtain patent licenses, in effect making the
program proprietary. To prevent this, we have made it clear that any
patent must be licensed for everyone's free use or not licensed at all.
The precise terms and conditions for copying, distribution and
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GNU GENERAL PUBLIC LICENSE
TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
0. This License applies to any program or other work which contains
a notice placed by the copyright holder saying it may be distributed
under the terms of this General Public License. The "Program", below,
refers to any such program or work, and a "work based on the Program"
means either the Program or any derivative work under copyright law:
that is to say, a work containing the Program or a portion of it,
either verbatim or with modifications and/or translated into another
language. (Hereinafter, translation is included without limitation in
the term "modification".) Each licensee is addressed as "you".
Activities other than copying, distribution and modification are not
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is covered only if its contents constitute a work based on the
Program (independent of having been made by running the Program).
Whether that is true depends on what the Program does.
1. You may copy and distribute verbatim copies of the Program's
source code as you receive it, in any medium, provided that you
conspicuously and appropriately publish on each copy an appropriate
copyright notice and disclaimer of warranty; keep intact all the
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and give any other recipients of the Program a copy of this License
along with the Program.
You may charge a fee for the physical act of transferring a copy, and
you may at your option offer warranty protection in exchange for a fee.
2. You may modify your copy or copies of the Program or any portion
of it, thus forming a work based on the Program, and copy and
distribute such modifications or work under the terms of Section 1
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a) You must cause the modified files to carry prominent notices
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b) You must cause any work that you distribute or publish, that in
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c) If the modified program normally reads commands interactively
when run, you must cause it, when started running for such
interactive use in the most ordinary way, to print or display an
announcement including an appropriate copyright notice and a
notice that there is no warranty (or else, saying that you provide
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the Program is not required to print an announcement.)
These requirements apply to the modified work as a whole. If
identifiable sections of that work are not derived from the Program,
and can be reasonably considered independent and separate works in
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Thus, it is not the intent of this section to claim rights or contest
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collective works based on the Program.
In addition, mere aggregation of another work not based on the Program
with the Program (or with a work based on the Program) on a volume of
a storage or distribution medium does not bring the other work under
the scope of this License.
3. You may copy and distribute the Program (or a work based on it,
under Section 2) in object code or executable form under the terms of
Sections 1 and 2 above provided that you also do one of the following:
a) Accompany it with the complete corresponding machine-readable
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customarily used for software interchange; or,
c) Accompany it with the information you received as to the offer
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distribution of the source code, even though third parties are not
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except as expressly provided under this License. Any attempt
otherwise to copy, modify, sublicense or distribute the Program is
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You are not responsible for enforcing compliance by third parties to
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infringement or for any other reason (not limited to patent issues),
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license would not permit royalty-free redistribution of the Program by
all those who receive copies directly or indirectly through you, then
the only way you could satisfy both it and this License would be to
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If any portion of this section is held invalid or unenforceable under
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It is not the purpose of this section to induce you to infringe any
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such claims; this section has the sole purpose of protecting the
integrity of the free software distribution system, which is
implemented by public license practices. Many people have made
generous contributions to the wide range of software distributed
through that system in reliance on consistent application of that
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to distribute software through any other system and a licensee cannot
impose that choice.
This section is intended to make thoroughly clear what is believed to
be a consequence of the rest of this License.
8. If the distribution and/or use of the Program is restricted in
certain countries either by patents or by copyrighted interfaces, the
original copyright holder who places the Program under this License
may add an explicit geographical distribution limitation excluding
those countries, so that distribution is permitted only in or among
countries not thus excluded. In such case, this License incorporates
the limitation as if written in the body of this License.
9. The Free Software Foundation may publish revised and/or new versions
of the General Public License from time to time. Such new versions will
be similar in spirit to the present version, but may differ in detail to
address new problems or concerns.
Each version is given a distinguishing version number. If the Program
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either of that version or of any later version published by the Free
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Foundation.
10. If you wish to incorporate parts of the Program into other free
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of preserving the free status of all derivatives of our free software and
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NO WARRANTY
11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN
OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
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WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
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POSSIBILITY OF SUCH DAMAGES.
END OF TERMS AND CONDITIONS
How to Apply These Terms to Your New Programs
If you develop a new program, and you want it to be of the greatest
possible use to the public, the best way to achieve this is to make it
free software which everyone can redistribute and change under these terms.
To do so, attach the following notices to the program. It is safest
to attach them to the start of each source file to most effectively
convey the exclusion of warranty; and each file should have at least
the "copyright" line and a pointer to where the full notice is found.
<one line to give the program's name and a brief idea of what it does.>
Copyright (C) <year> <name of author>
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License along
with this program; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
Also add information on how to contact you by electronic and paper mail.
If the program is interactive, make it output a short notice like this
when it starts in an interactive mode:
Gnomovision version 69, Copyright (C) year name of author
Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
This is free software, and you are welcome to redistribute it
under certain conditions; type `show c' for details.
The hypothetical commands `show w' and `show c' should show the appropriate
parts of the General Public License. Of course, the commands you use may
be called something other than `show w' and `show c'; they could even be
mouse-clicks or menu items--whatever suits your program.
You should also get your employer (if you work as a programmer) or your
school, if any, to sign a "copyright disclaimer" for the program, if
necessary. Here is a sample; alter the names:
Yoyodyne, Inc., hereby disclaims all copyright interest in the program
`Gnomovision' (which makes passes at compilers) written by James Hacker.
<signature of Ty Coon>, 1 April 1989
Ty Coon, President of Vice
This General Public License does not permit incorporating your program into
proprietary programs. If your program is a subroutine library, you may
consider it more useful to permit linking proprietary applications with the
library. If this is what you want to do, use the GNU Lesser General
Public License instead of this License.

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CC = gcc
CFLAGS = -Wall
LIBS = -lcrypto -lev
OBJ=pipe.o netsocket.o
BIN=pipe
all: logger pipe fmtsub_test
pipe: $(OBJ)
$(CC) $(CFLAGS) -o $(BIN) $(OBJ) $(EXTOBJ) $(LIBS)
logger: logger.o logger_test.o misc.o
$(CC) $(CFLAGS) -o logger logger.o logger_test.o misc.o
fmtsub_test: misc.o fmtsub_test.o
$(CC) $(CFLAGS) -o fmtsub_test misc.o fmtsub_test.o
doc:
doxygen
@echo ""
@echo "DOC URL: file://"`pwd`"/html/index.html"
@echo ""
clean:
rm -rf $(OBJ) $(BIN) logger fmtsub_test core *.o html/ latex/ man/
# Függőségek
#policy.o: server.h

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@ -0,0 +1,15 @@
Ring Bufferrel megoldhato a mem logging?
http://en.wikipedia.org/wiki/Circular_buffer
LIBEV STRICT ALIASING
=====================
netsocket.c -ben a libev örjöng:
* libjsi/netsocket.c: In function netsocket_connect:
* libjsi/netsocket.c:264: warning: dereferencing type-punned pointer will break strict-aliasing rules
A megoldás egyelőre az, hogy -fno-strict-aliasing CFLAGS-et használjuk.

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@ -0,0 +1,28 @@
# JSS C Library
This library contains the C functions and macros, which are often used in our
projects. In the future, this will be a stand-alone library with unit tests and
documentation.
[AMISMS](https://github.com/andrewjsi/amisms) and [libamievent](https://github.com/andrewjsi/libamievent) uses this library.
## Source code
The source code is well written as far as possible. We do not use tabs, instead
of 4 spaces is indented. All identifiers, including variables, function names
and macros written in English, but some comments and commits in Hungarian is,
because we are speaking and thinking in Hungarian. Nevertheless, we try to
write everything in English in the future.
## Contribution
It is an open source project, which is to be better and better. If you have any
ideas or find an error, or get stuck, you can contact us, please file a bug
report or send a pull-request!
## License
[_GPL2_](https://www.gnu.org/licenses/gpl-2.0.html)
(C) Copyright 2010-2014 Andras Jeszenszky, [JSS & Hayer
IT](http://www.jsshayer.hu) All Rights Reserved.

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CC = gcc
CFLAGS = -Wall -ggdb -I..
CFLAGS += -DCON_DEBUG
LIBS = -lev -lm -lmysqlclient -lpthread
OBJ=amysql.o ../logger.o ../misc.o
all: test_amysql test_sync_amysql
test_amysql: test_amysql.o $(OBJ)
$(CC) $(CFLAGS) -o test_amysql test_amysql.o $(OBJ) $(LIBS)
test_sync_amysql: test_sync_amysql.o $(OBJ)
$(CC) $(CFLAGS) -o test_sync_amysql test_sync_amysql.o $(OBJ) $(LIBS)
clean:
rm -f $(OBJ) *.o test_amysql test_sync_amysql

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libzdb - http://www.tildeslash.com/libzdb/
A small, easy to use Open Source Database Connection Pool Library with the
following features:
-Thread safe Database Connection Pool
-Connect to multiple database systems
-Zero runtime configuration, connect using a URL scheme
-Supports MySQL, PostgreSQL, SQLite and Oracle
MySQL C Library API Reference:
http://dev.mysql.com/doc/refman/5.1/en/c-api-functions.html
MySQL C API Tutorial:
http://zetcode.com/tutorials/mysqlcapitutorial/
MySQL szerverben trigger - adding a New User-Defined Function
http://dev.mysql.com/doc/refman/5.5/en/adding-udf.html
http://jan.kneschke.de/2008/9/9/async-mysql-queries-with-c-api/
tömör leírás az async elvről:
http://betterlogic.com/roger/2008/08/gdb-says-asynchronous-mysql-c-api/
http://stackoverflow.com/questions/11313686/using-libmysqlclient-in-multi-threaded-application
http://www.linuxtopia.org/online_books/database_guides/mysql_5.1_database_reference_guide/threaded-clients.html
http://dev.mysql.com/doc/refman/5.1/en/threaded-clients.html

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/* amysql.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/* TODO: megvizsgálni a libzdb-t, mert ismeri a MySQL-t, PostgreSQL-t és az
Oracle-t is. A libzdb tudtommal nem aszinkron, viszont az amysql.c mintájára
aszinkronná lehetne tenni:) */
#include <stdarg.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <pthread.h>
#include <mysql/mysql.h>
#include <ev.h>
#include "debug.h"
#include "logger.h"
#define FIELDBUFSIZE 8192
#define FIELDVSIZE 32
#define QUERYBUFSIZE 4096
MYSQL *db;
MYSQL_RES *result;
MYSQL_ROW row;
int busy = 0;
void (*callback)(int, char**, char*);
int fieldc; // result mezők száma
char *fieldv[FIELDVSIZE]; // result mező elemek
char fieldbuf[FIELDBUFSIZE]; // result buffer (ide mutatnak a fieldv elemek)
char querybuf[QUERYBUFSIZE]; // itt tároljuk el a query-t
char error_str[256];
int error_num;
struct option_t {
char host[64];
char user[64];
char password[64];
char database[64];
char charset[16];
int connect_timeout;
int data_timeout;
} option = {
.host = "localhost",
.user = "",
.password = "",
.database = "",
.charset = "", // alapértelmezetten nem piszkáljuk a charset-et
.connect_timeout = 10, // TODO: prod környezetben ezeket magasra venni!
.data_timeout = 20, /* Each attempt uses this timeout value and there
are retries if necessary, so the total effective
timeout value is three times the option value. */
};
static ev_async evasync_main;
static ev_async evasync_thread;
static struct ev_loop *loop_main; // főprogram event loopja (kapja az amysql_init())
static struct ev_loop *loop_thread; // thread event loopja
pthread_t thread_id;
void amysql_option_host (const char *host) {
strncpy(option.host, host, sizeof(option.host) - 1);
}
void amysql_option_user (const char *user) {
strncpy(option.user, user, sizeof(option.user) - 1);
}
void amysql_option_password (const char *password) {
strncpy(option.password, password, sizeof(option.password) - 1);
}
void amysql_option_database (const char *database) {
strncpy(option.database, database, sizeof(option.database) - 1);
}
void amysql_option_charset (const char *charset) {
strncpy(option.charset, charset, sizeof(option.charset) - 1);
}
void amysql_option_connect_timeout (int connect_timeout) {
option.connect_timeout = connect_timeout;
}
void amysql_option_data_timeout (int data_timeout) {
option.data_timeout = data_timeout;
}
// thread: AMYSQL, ha amysql_query() hívás volt
// thread: nincs definiálva, ha amysql_sync_query() hívás volt
// Függvény a következőket csinálja:
// reset, init, config, connect, query, fetch, save, disconnect, free
static void query () {
con_debug("send query: %s", querybuf);
//** Buffer reset **//
memset(fieldbuf, 0, sizeof(fieldbuf));
memset(fieldv, 0, sizeof(fieldv));
memset(error_str, 0, sizeof(error_str));
fieldc = 0;
db = NULL;
result = NULL;
//** Init **//
db = mysql_init(NULL);
if (db == NULL) {
con_debug("mysql_init() returned NULL");
char *errmsg = "mysql_init(): Can't allocate memory";
strncpy(error_str, errmsg, sizeof(error_str));
error_num = -1;
goto done;
}
//** Config **//
mysql_options(db, MYSQL_OPT_CONNECT_TIMEOUT, &option.connect_timeout);
mysql_options(db, MYSQL_OPT_READ_TIMEOUT, &option.data_timeout);
mysql_options(db, MYSQL_OPT_WRITE_TIMEOUT, &option.data_timeout);
//** Connect **//
if (!mysql_real_connect(db, option.host, option.user, option.password, option.database, 0, NULL, 0)) {
con_debug("Can't connect to MySQL server: %s", mysql_error(db));
strncpy(error_str, mysql_error(db), sizeof(error_str));
error_num = -1;
goto done;
}
if (option.charset && strlen(option.charset)) {
if (mysql_set_character_set(db, option.charset)) {
strncpy(error_str, mysql_error(db), sizeof(error_str));
error_num = -1;
goto done;
}
}
//** Query **//
if (mysql_query(db, querybuf)) {;
con_debug("query failed: %s", mysql_error(db));
strncpy(error_str, mysql_error(db), sizeof(error_str));
error_num = -1;
goto done;
}
//** Result **//
result = mysql_store_result(db);
if (result == NULL) { // INSERT INTO sikeres
con_debug("INSERT or UPDATE OK: result = NULL");
error_num = 0;
goto done;
}
//** Fetch only ONE row **//
int num_fields = mysql_num_fields(result);
row = mysql_fetch_row(result);
if (row == NULL) { // SELECT-ben a WHERE nem illeszkedett semmire
con_debug("SELECT empty: row = NULL");
error_num = 0;
goto done;
}
/* Az alábbi for ciklus a következőket csinálja: Egyetlen row mezőinek
másolása fieldbuf bufferbe és feldarabolása fieldv-be. A fieldbuf-ba
folyamatosan bemásoljuk a row-ból kiolvasott byteokat. Ezzel a
folytonossággal helyet spórolunk meg, mert nem kell minden mezőnek fix
méretet lefoglalni. A fieldv mutató tömbbe pedig az egyes mezők
kezdőcimeit mentjük el, ami a fieldbuf azon indexére mutat, ahol az
adott mező kezdődik. Az argc és *argv[] pároshoz hasonlóan a fieldc
változóba kerül a fieldv tömb mérete. Ha elegendő volt a FIELDBUFSIZE és
a FIELDVSIZE, akkor ez megegyezik a MySQL által visszaadott
mysql_num_fields() értékével. Ellenkező esetben annyi lesz, amennyit
sikerült kigyűjteni, tehát amennyit biztonsággal ki lehet olvasni. */
//** Save **//
int rpos = 0;
for (fieldc = 0; fieldc < num_fields && fieldc < FIELDVSIZE; fieldc++) {
if (row[fieldc] == NULL) {
fieldv[fieldc] = NULL;
} else {
fieldv[fieldc] = &fieldbuf[rpos];
int i;
for (i = 0; row[fieldc][i] != 0; i++) {
fieldbuf[rpos] = row[fieldc][i];
rpos++;
if (rpos > FIELDBUFSIZE - 2) { // ne szaladjunk túl a fieldbuf méretén
fieldc++;
error_num = 0;
goto done;
}
}
fieldbuf[rpos] = 0;
rpos++;
if (rpos > FIELDBUFSIZE - 2) { // ne szaladjunk túl a fieldbuf méretén
fieldc++;
error_num = 0;
goto done;
}
}
}
//~ int k; for (k = 0; k < fieldc; k++) printf("%d s %s s\n", k, fieldv[k]); printf("\n");
done:
//** Free & Disconnect **//
if (result != NULL)
mysql_free_result(result);
if (db != NULL)
mysql_close(db);
// TODO: strerror üzenetét hozzácsapni az errstr-hez, az err != 0
// ha 4-es hiba (interrupted system call) jön, akkor azt a mysql_connect()
// timeout okozza. Vajon ez hogy van megoldva? SIGALRM??? Az bebaszna!
//perror("strerror()");
}
void query_and_evasync () {
query();
con_debug("ev_async: AMYSQL -> MAIN");
ev_async_send(loop_main, &evasync_main);
}
// thread: MAIN
int amysql_is_busy () {
return busy;
}
// thread: MAIN
int amysql_query (void *cb, const char *fmt, ...) {
if (busy) {
con_debug("amysql is BUSY");
return -1;
}
busy = 1;
va_list ap;
va_start(ap, fmt);
vsnprintf(querybuf, sizeof(querybuf) - 1, fmt, ap);
va_end(ap);
callback = cb;
con_debug("ev_async: MAIN -> AMYSQL");
ev_async_send(loop_thread, &evasync_thread);
return 0;
}
// thread: MAIN
static void invoke_callback (EV_P_ ev_async *w, int revents) {
/* callback hívás közben már lehet új amysql_query() függvényt hívni,
mert féligmeddig biztonságos. Lehetőség szerint a callback függvény
legvégén legyen új amysql_query() hívás, mert az amysql thread felül
fogja írni a fieldc/fieldv és error_num/error_str változókat. */
busy = 0;
callback(fieldc, fieldv, (error_num) ? error_str : NULL);
}
// thread: AMYSQL
static void *infinite_loop () {
/* Itt lehet inicializálni az AMYSQL thread saját event watchereit */
ev_loop(loop_thread, 0);
con_debug("FATAL: amysql event loop exited! ");
return NULL;
}
// thread: MAIN
int amysql_init (struct ev_loop *loop) {
busy = 0;
loop_main = loop;
loop_thread = ev_loop_new(EVFLAG_AUTO);
ev_async_init(&evasync_main, (void*)invoke_callback);
ev_async_init(&evasync_thread, (void*)query_and_evasync);
/* A mostani kezdetleges thread tapasztalataim alapján, szerintem az
ev_async watchereket itt, még a pthread_create() előtt kell indítani.
Azaz még a hívó thread kontextusában. Mert ha az új thread-be
indítanánk, akkor elképzelhető, hogy visszatérés után a hívó függvény
egyből egy ev_async_send() hívással folytatja a futást és lehet, hogy az
új thread még nem járna az ev_async_init()-nél és ev_async_start()-nál.
Ez pedig ugye nem , mert azelőtt hívnánk meg valamit, mint hogy
inicializáltuk volna. A többi, (a thread-re nézve saját) event watcher,
io, időzítő, stb.. inicializálását már végezheti a thread saját maga,
tehát mehet a pthread_create() által meghívott függvénybe. */
ev_async_start(loop_main, &evasync_main);
ev_async_start(loop_thread, &evasync_thread);
int err;
err = pthread_create(&thread_id, NULL, infinite_loop, NULL);
if (err)
con_debug("thread creating error: (%d) %s", err, strerror(err));
return err;
}
int amysql_sync_query (int *parc, char **parv[], const char *fmt, ...) {
va_list ap;
va_start(ap, fmt);
vsnprintf(querybuf, sizeof(querybuf) - 1, fmt, ap);
va_end(ap);
query();
if (parc)
*parc = fieldc;
if (parv)
*parv = fieldv;
return error_num;
}
const char *amysql_strerror () {
return (error_num) ? error_str : NULL;
}

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/* amysql.h
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#include <ev.h>
/* Azért kell a loop paraméter, mert innen tudja az amysql, hogy melyik
event loop-ban kell majd meghívni a callback-et.*/
int amysql_init (struct ev_loop *loop);
/* Az amysql_query() azonnal visszaadja a vezérlést, és a háttérben egy
másik szálon megkezdi a paraméterben megadott lekérdezést az adatbázis
szerver felé. Amikor megjön a válasz, akkor lezárja a szerverrel a
kapcsolatot és meghívja a paraméterben megadott callback függvényt.
Figyelem! A callback függvény végén, de tényleg a legvégén lehetőség van új
amysql_query() függvény hívására. Hatására a fieldc/fieldv és a hibaváltozók
törlődni fognak, amiket a callback függvény is kapott.*/
int amysql_query (void *cb, const char *fmt, ...);
/* Konfigurációs függvények */
// default: localhost
void amysql_option_host (const char *host);
void amysql_option_user (const char *user);
void amysql_option_password (const char *password);
void amysql_option_database (const char *database);
// default: nem hívja meg a mysql_set_character_set() függvényt
void amysql_option_charset (const char *charset);
// default: 10
void amysql_option_connect_timeout (int connect_timeout);
// default: 20
void amysql_option_data_timeout (int data_timeout);
// visszaadja a hiba szövegét vagy NULL-t, ha nem volt hiba
const char *amysql_strerror ();
// nem-aszinkron lekérdezés. A lekérdezés erejéig az amysql_sync_query() hívás
// blokkol. Az eredményt a parc és a parv-be menti el.
int amysql_sync_query (int *parc, char **parv[], const char *fmt, ...);

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/* test_amysql.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#include <stdarg.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <ev.h>
#include "amysql.h"
#include "debug.h"
static struct ev_loop *loop;
static ev_timer timeout_watcher;
static void cbtest (int fc, char *fv[], char *e);
static void q () {
amysql_query(cbtest, "select * from smsout where sent='0' order by RAND()");
}
static void cbtest (int fc, char *fv[], char *e) {
debi(fc);
printf("e=%s\n", e);
int i;
for (i = 0; i < fc; i++) {
printf("%d --- %s\n", i, fv[i]);
}
// 3 másodperc múlva q() futtatása
ev_once(loop, -1, -1, 3, q, 0);
}
static void timeout_cb (EV_P_ ev_timer *w, int revents) {
printf("timeout\n");
}
int main (int argc, char *argv[]) {
loop = ev_default_loop (0);
ev_timer_init (&timeout_watcher, timeout_cb, 0, 0.5);
ev_timer_start (loop, &timeout_watcher);
amysql_init(loop);
q(); // berúgjuk a mocit:)
ev_loop (loop, 0);
return 0;
}

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/* test_sync_amysql.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#include <stdarg.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "amysql.h"
#include "debug.h"
int main (int argc, char *argv[]) {
amysql_option_user ("proba");
amysql_option_password ("proba");
amysql_option_database ("proba");
amysql_option_charset ("utf8");
// if (amysql_sync_query(NULL, NULL, "INSERT INTO proba (name, comment) VALUES ('%s', '%s')", "egy", "kettő")) {
// printf("ERROR: %s\n", amysql_strerror());
// }
int parc;
char **parv;
if (amysql_sync_query(&parc, &parv, "select * from proba order by RAND()")) {
printf("ERROR: %s\n", amysql_strerror());
}
int i;
for (i = 0; i < parc; i++) {
printf("%d --- %s\n", i, parv[i]);
}
return 0;
}

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/* debug.h
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#define debi(x) printf(#x " = %d\n", ((int)(x)))
#define debf(x) printf(#x " = %f\n", (x))
#define debs(x) printf(#x " = %s\n", (x))

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/* fmtsub_test.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include "misc.h"
/*
const char *args[][2] = {
{"%n", "2463434"},
{"%v", "0.9.6-rc2"},
{"%n", "Jekusa Poromeck"},
{0}
};
*/
//~ void prob (const char *args[][2]) {
//~ int i;
//~ for (i = 0; afmt[i][0] != 0; i++) {
//~ printf("hello %s --- %s\n", afmt[i][0], afmt[i][1]);
//~
//~ }
//~ }
int main (int argc, char **argv) {
char buf2[32];
char buf[64];
char *pattern = argv[1];
char proba[64];
strcpy(proba, "PROBA");
const char *args[][2] = {
{"%n", NULL},
{"", "URES_STRING"},
{"%v", "0.9.6-rc2"},
{"%a", "Jekusa Poromeck"},
{"%p", proba},
{"Windows", "Linux"},
{"\\%", "%"},
{0}
};
if (fmtsub(buf, sizeof(buf), pattern, args)) {
printf("fmtsub() hibaval tert vissza!\n");
}
printf("%d s %s s\n", (int)strlen(buf), buf);
strncpy(buf2, buf, sizeof(buf2)-1);
printf("%d s %s s\n", (int)strlen(buf2), buf2);
return 0;
}

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CC = gcc
CFLAGS = -Wall -I.. -ggdb
LDFLAGS = -lpthread
all: test_htclient
test_htclient: test_htclient.o htclient.o url_parser.o
$(CC) $(CFLAGS) -o test_htclient test_htclient.o htclient.o url_parser.o $(LDFLAGS)
clean:
rm -f test_htclient *.o

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/* htclient.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <string.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <netdb.h>
#include <stdarg.h>
#include "htclient.h"
#include "debug.h"
#include "url_parser.h"
/* void ERR(...)
*
* A paraméterül átadott VA_ARGS formátumú hibaüzenetet bemásolja a req->error
* -ba, majd visszatér return -1 -el. Figyelni kell a blokkos szerkezeteknél:
* Mindenképp használni kell a { } zárójeleket, mivel ez a makró két utasítást
* tartalmaz. A { } elhagyásával logikai hiba keletkezhet a programban.
*/
#define ERR(...) snprintf(htc->error, sizeof(htc->error), __VA_ARGS__); return -1
#define ARRAY_SIZE(array) (sizeof(array) / sizeof(array[0]))
void htclient_dump (htclient_t *htc) {
#define HTCLIENT_DUMPS(s) printf("%13s = %s\n", #s, htc->s)
#define HTCLIENT_DUMPI(i) printf("%13s = %d\n", #i, htc->i)
HTCLIENT_DUMPS(url.full);
HTCLIENT_DUMPS(url.scheme);
HTCLIENT_DUMPS(url.host);
HTCLIENT_DUMPS(url.port);
HTCLIENT_DUMPS(url.path);
HTCLIENT_DUMPS(url.query);
HTCLIENT_DUMPS(url.fragment);
HTCLIENT_DUMPS(url.username);
HTCLIENT_DUMPS(url.password);
HTCLIENT_DUMPS(request_data);
HTCLIENT_DUMPS(response_data);
HTCLIENT_DUMPI(num_headers);
HTCLIENT_DUMPS(header_buf);
HTCLIENT_DUMPS(error);
}
// Skip the characters until one of the delimiters characters found.
// 0-terminate resulting word. Skip the rest of the delimiters if any. Advance
// pointer to buffer to the next word. Return found 0-terminated word.
// (function from Mongoose project)
static char *skip (char **buf, const char *delimiters) {
char *p, *begin_word, *end_word, *end_delimiters;
begin_word = *buf;
end_word = begin_word + strcspn(begin_word, delimiters);
end_delimiters = end_word + strspn(end_word, delimiters);
for (p = end_word; p < end_delimiters; p++) {
*p = '\0';
}
*buf = end_delimiters;
return begin_word;
}
// Parse HTTP headers from the given buffer, advance buffer to the point where
// parsing stopped. (function from Mongoose project)
static void parse_http_headers(char **buf, struct htclient_t *htc) {
size_t i;
for (i = 0; i < ARRAY_SIZE(htc->headers); i++) {
htc->headers[i].name = skip(buf, ": ");
htc->headers[i].value = skip(buf, "\r\n");
if (htc->headers[i].name[0] == '\0')
break;
htc->num_headers = i + 1;
}
}
char *htclient_error (htclient_t *htc) {
if (htc->error == NULL || !strlen(htc->error))
return NULL;
return htc->error;
}
int htclient_url (htclient_t *htc, const char *fmt, ...) {
va_list ap;
va_start(ap, fmt);
vsnprintf(htc->url.full, sizeof(htc->url.full), fmt, ap);
va_end(ap);
struct parsed_url *pu = parse_url(htc->url.full);
if (pu == NULL) {
htc->url.set = 0;
ERR("URL mismatch");
}
htc->url.url_parser_ptr = pu;
htc->url.scheme = pu->scheme;
htc->url.host = pu->host;
htc->url.port = pu->port;
htc->url.path = pu->path;
htc->url.query = pu->query;
htc->url.fragment = pu->fragment;
htc->url.username = pu->username;
htc->url.password = pu->password;
// ha nincs megadva a port, akkor a scheme-ből megállapítjuk
if (!htc->url.port && htc->url.scheme) {
if (!strcmp(htc->url.scheme, "http")) {
// TODO: az strdup() nem okoz memória szivárgást? Ha felszabadítjuk
// az url_parser-t, akkor az felszabadítja ezt is?
htc->url.port = strdup("80");
} else if (!strcmp(htc->url.scheme, "https")) {
htc->url.port = strdup("443");
}
}
// ha valamilyen oknál fogva ezen a ponton nincs
// host vagy nincs port, akkor a további
// grimbuszok elkerülése végett hivát dobunk:)
if (!htc->url.host || !htc->url.port) {
htc->url.set = 0;
ERR("URL mismatch");
}
htc->url.set = 1;
return 0;
}
int htclient_perform (htclient_t *htc) {
struct sockaddr_in serv_addr;
struct hostent *server;
int port;
int sockfd;
int rv;
if (!htc->url.set) {
ERR("URL not set or bogus");
}
port = atoi(htc->url.port);
sockfd = socket(AF_INET, SOCK_STREAM, 0);
if (sockfd < 0) {
ERR("error opening socket: %s", strerror(errno));
}
server = gethostbyname(htc->url.host);
if (server == NULL) {
close(sockfd);
ERR("host not found: %s", htc->url.host);
}
bzero((char *) &serv_addr, sizeof(serv_addr));
serv_addr.sin_family = AF_INET;
bcopy((char *)server->h_addr,
(char *)&serv_addr.sin_addr.s_addr,
server->h_length);
serv_addr.sin_port = htons(port);
if (connect(sockfd, (struct sockaddr*) &serv_addr, sizeof(serv_addr)) < 0) {
close(sockfd);
ERR("can't connect to %s:%d: %s", htc->url.host, port, strerror(errno));
}
rv = write(sockfd, htc->request_data, strlen(htc->request_data));
if (rv < 0) {
close(sockfd);
ERR("ERROR writing to socket");
}
memset(htc->response_data, 0, sizeof(htc->response_data));
rv = read(sockfd, htc->response_data, sizeof(htc->response_data) - 1);
if (rv < 0) {
close(sockfd);
ERR("ERROR reading from socket");
}
htc->header_buf = strdup(htc->response_data);
char *buf = htc->header_buf;
parse_http_headers(&buf, htc);
close(sockfd);
return 0;
}
char *htclient_header_get (htclient_t *htc, const char *name) {
int i;
for (i = 0; i < htc->num_headers; i++) {
if (htc->headers[i].name == NULL || htc->headers[i].value == NULL)
continue;
if (!strcmp(htc->headers[i].name, name))
return htc->headers[i].value;
}
return NULL;
}
htclient_t *htclient_new () {
htclient_t *htc = malloc(sizeof(*htc));
if (htc == NULL) {
fprintf(stderr, "out of memory");
return NULL;
}
memset(htc, 0, sizeof(*htc));
return htc;
}
void htclient_request_set (htclient_t *htc, const char *fmt, ...) {
va_list ap;
va_start(ap, fmt);
vsnprintf(htc->request_data, sizeof(htc->request_data), fmt, ap);
va_end(ap);
}
void htclient_destroy (htclient_t *htc) {
if (htc == NULL)
return;
if (htc->header_buf != NULL) {
free(htc->header_buf);
htc->header_buf = NULL;
}
free(htc);
}

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/* htclient.h
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#define HTCLIENT_BUFSIZ 1024
// változás esetén módosítsd a htclient_dump() függvényt a htclient.c fájlban!
typedef struct htclient_t {
struct {
void *url_parser_ptr; // url_parser.h -ban definiált pointer, ezt kell majd felszabadítani
char full[HTCLIENT_BUFSIZ]; // http://jsi@jss.hu/auth?param=59
char *scheme; // http
char *host; // jss.hu
char *port; // (null)
char *path; // auth
char *query; // param=59
char *fragment; // (null)
char *username; // jsi
char *password; // (null)
int set; // 1 lesz, ha rendben beállítottuk az URL-t
} url;
char request_data[HTCLIENT_BUFSIZ];
char response_data[HTCLIENT_BUFSIZ];
struct {
char *name;
char *value;
} headers[16];
int num_headers;
char *header_buf;
char error[128];
} htclient_t;
// új htclient objektum
htclient_t *htclient_new ();
// URL beállítása
// Kezeli a scheme, host, port, path URL részeket. Tovább infó a
// htclient_t->url struktúrában. Az URL path részét nem kezeli a
// htclient_perform(), ezért azt külön el kell menteni a request-ben.
//
// például: http://jss.hu:3333/akarmi.jpg
int htclient_url (htclient_t *htc, const char *fmt, ...);
// Beállítja a teljes HTTP request-et. Beleértve a fejléceket meg mindent.
void htclient_request_set (htclient_t *htc, const char *fmt, ...);
// Végrehajtja a lekérdezést. A függvény addig blokkol, amíg nem érkezik meg a
// válasz vagy valami hiba történik.
int htclient_perform (htclient_t *htc);
// Visszaadja a válaszban szereplő header értékét. Ha a megadott header nem
// létezik, akkor a visszatérési érték NULL.
char *htclient_header_get (htclient_t *htc, const char *name);
// Felszabadítja a htclient objektumot
void htclient_destroy (htclient_t *htc);
// Ha volt hiba, akkor visszaadja a hibastringet, ha nem, akkor NULL
char *htclient_error (htclient_t *htc);
// htclient_t struktúra dumpolása
void htclient_dump (htclient_t *htc);

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/* test_htclient.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <string.h>
#include "htclient.h"
#include "debug.h"
#include "url_parser.h"
void url_test () {
htclient_t *h = htclient_new();
htclient_url(h, "http://10.42.20.14:3000/akarmi.jpg");
htclient_dump(h);
htclient_destroy(h);
}
int main (int argc, const char *argv[]) {
// url_test(); return(0);
htclient_t *h;
h = htclient_new();
if (h == NULL) {
printf("out of memory\n");
return -1;
}
htclient_url(h, "http://10.42.20.14:3000/dovecheck");
htclient_request_set(h,
"GET /auth\r\n"
"Auth-server: akarmi\r\n"
"Auth-port: %d\r\n"
"\r\n"
, 5555
);
htclient_perform(h);
htclient_dump(h);
if (htclient_error(h)) {
printf("error: %s\n", htclient_error(h));
goto err;
}
char *hdr = htclient_header_get(h, "Auth-server");
printf("Header vissza: %s\n", hdr);
printf("len = %s\n", htclient_header_get(h, "Content-Length"));
htclient_destroy(h);
return 0;
err:
htclient_destroy(h);
return -1;
}

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/*_
* Copyright 2010-2011 Scyphus Solutions Co. Ltd. All rights reserved.
*
* Authors:
* Hirochika Asai
*/
#include "url_parser.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <ctype.h>
/*
* Prototype declarations
*/
static __inline__ int _is_scheme_char(int);
/*
* Check whether the character is permitted in scheme string
*/
static __inline__ int
_is_scheme_char(int c)
{
return (!isalpha(c) && '+' != c && '-' != c && '.' != c) ? 0 : 1;
}
/*
* See RFC 1738, 3986
*/
struct parsed_url *
parse_url(const char *url)
{
struct parsed_url *purl;
const char *tmpstr;
const char *curstr;
int len;
int i;
int userpass_flag;
int bracket_flag;
/* Allocate the parsed url storage */
purl = malloc(sizeof(struct parsed_url));
if ( NULL == purl ) {
return NULL;
}
purl->scheme = NULL;
purl->host = NULL;
purl->port = NULL;
purl->path = NULL;
purl->query = NULL;
purl->fragment = NULL;
purl->username = NULL;
purl->password = NULL;
curstr = url;
/*
* <scheme>:<scheme-specific-part>
* <scheme> := [a-z\+\-\.]+
* upper case = lower case for resiliency
*/
/* Read scheme */
tmpstr = strchr(curstr, ':');
if ( NULL == tmpstr ) {
/* Not found the character */
parsed_url_free(purl);
return NULL;
}
/* Get the scheme length */
len = tmpstr - curstr;
/* Check restrictions */
for ( i = 0; i < len; i++ ) {
if ( !_is_scheme_char(curstr[i]) ) {
/* Invalid format */
parsed_url_free(purl);
return NULL;
}
}
/* Copy the scheme to the storage */
purl->scheme = malloc(sizeof(char) * (len + 1));
if ( NULL == purl->scheme ) {
parsed_url_free(purl);
return NULL;
}
(void)strncpy(purl->scheme, curstr, len);
purl->scheme[len] = '\0';
/* Make the character to lower if it is upper case. */
for ( i = 0; i < len; i++ ) {
purl->scheme[i] = tolower(purl->scheme[i]);
}
/* Skip ':' */
tmpstr++;
curstr = tmpstr;
/*
* //<user>:<password>@<host>:<port>/<url-path>
* Any ":", "@" and "/" must be encoded.
*/
/* Eat "//" */
for ( i = 0; i < 2; i++ ) {
if ( '/' != *curstr ) {
parsed_url_free(purl);
return NULL;
}
curstr++;
}
/* Check if the user (and password) are specified. */
userpass_flag = 0;
tmpstr = curstr;
while ( '\0' != *tmpstr ) {
if ( '@' == *tmpstr ) {
/* Username and password are specified */
userpass_flag = 1;
break;
} else if ( '/' == *tmpstr ) {
/* End of <host>:<port> specification */
userpass_flag = 0;
break;
}
tmpstr++;
}
/* User and password specification */
tmpstr = curstr;
if ( userpass_flag ) {
/* Read username */
while ( '\0' != *tmpstr && ':' != *tmpstr && '@' != *tmpstr ) {
tmpstr++;
}
len = tmpstr - curstr;
purl->username = malloc(sizeof(char) * (len + 1));
if ( NULL == purl->username ) {
parsed_url_free(purl);
return NULL;
}
(void)strncpy(purl->username, curstr, len);
purl->username[len] = '\0';
/* Proceed current pointer */
curstr = tmpstr;
if ( ':' == *curstr ) {
/* Skip ':' */
curstr++;
/* Read password */
tmpstr = curstr;
while ( '\0' != *tmpstr && '@' != *tmpstr ) {
tmpstr++;
}
len = tmpstr - curstr;
purl->password = malloc(sizeof(char) * (len + 1));
if ( NULL == purl->password ) {
parsed_url_free(purl);
return NULL;
}
(void)strncpy(purl->password, curstr, len);
purl->password[len] = '\0';
curstr = tmpstr;
}
/* Skip '@' */
if ( '@' != *curstr ) {
parsed_url_free(purl);
return NULL;
}
curstr++;
}
if ( '[' == *curstr ) {
bracket_flag = 1;
} else {
bracket_flag = 0;
}
/* Proceed on by delimiters with reading host */
tmpstr = curstr;
while ( '\0' != *tmpstr ) {
if ( bracket_flag && ']' == *tmpstr ) {
/* End of IPv6 address. */
tmpstr++;
break;
} else if ( !bracket_flag && (':' == *tmpstr || '/' == *tmpstr) ) {
/* Port number is specified. */
break;
}
tmpstr++;
}
len = tmpstr - curstr;
purl->host = malloc(sizeof(char) * (len + 1));
if ( NULL == purl->host || len <= 0 ) {
parsed_url_free(purl);
return NULL;
}
(void)strncpy(purl->host, curstr, len);
purl->host[len] = '\0';
curstr = tmpstr;
/* Is port number specified? */
if ( ':' == *curstr ) {
curstr++;
/* Read port number */
tmpstr = curstr;
while ( '\0' != *tmpstr && '/' != *tmpstr ) {
tmpstr++;
}
len = tmpstr - curstr;
purl->port = malloc(sizeof(char) * (len + 1));
if ( NULL == purl->port ) {
parsed_url_free(purl);
return NULL;
}
(void)strncpy(purl->port, curstr, len);
purl->port[len] = '\0';
curstr = tmpstr;
}
/* End of the string */
if ( '\0' == *curstr ) {
return purl;
}
/* Skip '/' */
if ( '/' != *curstr ) {
parsed_url_free(purl);
return NULL;
}
curstr++;
/* Parse path */
tmpstr = curstr;
while ( '\0' != *tmpstr && '#' != *tmpstr && '?' != *tmpstr ) {
tmpstr++;
}
len = tmpstr - curstr;
purl->path = malloc(sizeof(char) * (len + 1));
if ( NULL == purl->path ) {
parsed_url_free(purl);
return NULL;
}
(void)strncpy(purl->path, curstr, len);
purl->path[len] = '\0';
curstr = tmpstr;
/* Is query specified? */
if ( '?' == *curstr ) {
/* Skip '?' */
curstr++;
/* Read query */
tmpstr = curstr;
while ( '\0' != *tmpstr && '#' != *tmpstr ) {
tmpstr++;
}
len = tmpstr - curstr;
purl->query = malloc(sizeof(char) * (len + 1));
if ( NULL == purl->query ) {
parsed_url_free(purl);
return NULL;
}
(void)strncpy(purl->query, curstr, len);
purl->query[len] = '\0';
curstr = tmpstr;
}
/* Is fragment specified? */
if ( '#' == *curstr ) {
/* Skip '#' */
curstr++;
/* Read fragment */
tmpstr = curstr;
while ( '\0' != *tmpstr ) {
tmpstr++;
}
len = tmpstr - curstr;
purl->fragment = malloc(sizeof(char) * (len + 1));
if ( NULL == purl->fragment ) {
parsed_url_free(purl);
return NULL;
}
(void)strncpy(purl->fragment, curstr, len);
purl->fragment[len] = '\0';
curstr = tmpstr;
}
return purl;
}
/*
* Free memory of parsed url
*/
void
parsed_url_free(struct parsed_url *purl)
{
if ( NULL != purl ) {
if ( NULL != purl->scheme ) {
free(purl->scheme);
}
if ( NULL != purl->host ) {
free(purl->host);
}
if ( NULL != purl->port ) {
free(purl->port);
}
if ( NULL != purl->path ) {
free(purl->path);
}
if ( NULL != purl->query ) {
free(purl->query);
}
if ( NULL != purl->fragment ) {
free(purl->fragment);
}
if ( NULL != purl->username ) {
free(purl->username);
}
if ( NULL != purl->password ) {
free(purl->password);
}
free(purl);
}
}
/*
* Local variables:
* tab-width: 4
* c-basic-offset: 4
* End:
* vim600: sw=4 ts=4 fdm=marker
* vim<600: sw=4 ts=4
*/

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/*_
* Copyright 2010 Scyphus Solutions Co. Ltd. All rights reserved.
*
* Authors:
* Hirochika Asai
*/
#ifndef _URL_PARSER_H
#define _URL_PARSER_H
/*
* URL storage
*/
struct parsed_url {
char *scheme; /* mandatory */
char *host; /* mandatory */
char *port; /* optional */
char *path; /* optional */
char *query; /* optional */
char *fragment; /* optional */
char *username; /* optional */
char *password; /* optional */
};
#ifdef __cplusplus
extern "C" {
#endif
/*
* Declaration of function prototypes
*/
struct parsed_url *parse_url (const char *);
void parsed_url_free (struct parsed_url *);
#ifdef __cplusplus
}
#endif
#endif /* _URL_PARSER_H */
/*
* Local variables:
* tab-width: 4
* c-basic-offset: 4
* End:
* vim600: sw=4 ts=4 fdm=marker
* vim<600: sw=4 ts=4
*/

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The "inih" library is distributed under the New BSD license:
Copyright (c) 2009, Brush Technology
All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
* Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
* Neither the name of Brush Technology nor the names of its contributors
may be used to endorse or promote products derived from this software
without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY BRUSH TECHNOLOGY ''AS IS'' AND ANY
EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL BRUSH TECHNOLOGY BE LIABLE FOR ANY
DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

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CFLAGS = -Wall
all: test_ini
test_ini: ini.o test_ini.o
gcc -o test_ini $(CFLAGS) test_ini.o ini.o
clean:
rm -f test_ini *.o

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inih is a simple .INI file parser written in C, released under the New BSD
license (see LICENSE.txt). Go to the project home page for more info:
http://code.google.com/p/inih/

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// Read an INI file into easy-to-access name/value pairs.
#include <algorithm>
#include <cctype>
#include <cstdlib>
#include "../ini.h"
#include "INIReader.h"
using std::string;
INIReader::INIReader(string filename)
{
_error = ini_parse(filename.c_str(), ValueHandler, this);
}
int INIReader::ParseError()
{
return _error;
}
string INIReader::Get(string section, string name, string default_value)
{
string key = MakeKey(section, name);
return _values.count(key) ? _values[key] : default_value;
}
long INIReader::GetInteger(string section, string name, long default_value)
{
string valstr = Get(section, name, "");
const char* value = valstr.c_str();
char* end;
// This parses "1234" (decimal) and also "0x4D2" (hex)
long n = strtol(value, &end, 0);
return end > value ? n : default_value;
}
double INIReader::GetReal(string section, string name, double default_value)
{
string valstr = Get(section, name, "");
const char* value = valstr.c_str();
char* end;
double n = strtod(value, &end);
return end > value ? n : default_value;
}
bool INIReader::GetBoolean(string section, string name, bool default_value)
{
string valstr = Get(section, name, "");
// Convert to lower case to make string comparisons case-insensitive
std::transform(valstr.begin(), valstr.end(), valstr.begin(), ::tolower);
if (valstr == "true" || valstr == "yes" || valstr == "on" || valstr == "1")
return true;
else if (valstr == "false" || valstr == "no" || valstr == "off" || valstr == "0")
return false;
else
return default_value;
}
string INIReader::MakeKey(string section, string name)
{
string key = section + "." + name;
// Convert to lower case to make section/name lookups case-insensitive
std::transform(key.begin(), key.end(), key.begin(), ::tolower);
return key;
}
int INIReader::ValueHandler(void* user, const char* section, const char* name,
const char* value)
{
INIReader* reader = (INIReader*)user;
string key = MakeKey(section, name);
if (reader->_values[key].size() > 0)
reader->_values[key] += "\n";
reader->_values[key] += value;
return 1;
}

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// Read an INI file into easy-to-access name/value pairs.
// inih and INIReader are released under the New BSD license (see LICENSE.txt).
// Go to the project home page for more info:
//
// http://code.google.com/p/inih/
#ifndef __INIREADER_H__
#define __INIREADER_H__
#include <map>
#include <string>
// Read an INI file into easy-to-access name/value pairs. (Note that I've gone
// for simplicity here rather than speed, but it should be pretty decent.)
class INIReader
{
public:
// Construct INIReader and parse given filename. See ini.h for more info
// about the parsing.
INIReader(std::string filename);
// Return the result of ini_parse(), i.e., 0 on success, line number of
// first error on parse error, or -1 on file open error.
int ParseError();
// Get a string value from INI file, returning default_value if not found.
std::string Get(std::string section, std::string name,
std::string default_value);
// Get an integer (long) value from INI file, returning default_value if
// not found or not a valid integer (decimal "1234", "-1234", or hex "0x4d2").
long GetInteger(std::string section, std::string name, long default_value);
// Get a real (floating point double) value from INI file, returning
// default_value if not found or not a valid floating point value
// according to strtod().
double GetReal(std::string section, std::string name, double default_value);
// Get a boolean value from INI file, returning default_value if not found or if
// not a valid true/false value. Valid true values are "true", "yes", "on", "1",
// and valid false values are "false", "no", "off", "0" (not case sensitive).
bool GetBoolean(std::string section, std::string name, bool default_value);
private:
int _error;
std::map<std::string, std::string> _values;
static std::string MakeKey(std::string section, std::string name);
static int ValueHandler(void* user, const char* section, const char* name,
const char* value);
};
#endif // __INIREADER_H__

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// Example that shows simple usage of the INIReader class
#include <iostream>
#include "INIReader.h"
int main()
{
INIReader reader("../examples/test.ini");
if (reader.ParseError() < 0) {
std::cout << "Can't load 'test.ini'\n";
return 1;
}
std::cout << "Config loaded from 'test.ini': version="
<< reader.GetInteger("protocol", "version", -1) << ", name="
<< reader.Get("user", "name", "UNKNOWN") << ", email="
<< reader.Get("user", "email", "UNKNOWN") << ", pi="
<< reader.GetReal("user", "pi", -1) << ", active="
<< reader.GetBoolean("user", "active", true) << "\n";
return 0;
}

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// CFG(section, name, default)
CFG(protocol, version, "0")
CFG(user, name, "Fatty Lumpkin")
CFG(user, email, "fatty@lumpkin.com")
#undef CFG

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/* ini.h example that simply dumps an INI file without comments */
#include <stdio.h>
#include <string.h>
#include "../ini.h"
static int dumper(void* user, const char* section, const char* name,
const char* value)
{
static char prev_section[50] = "";
if (strcmp(section, prev_section)) {
printf("%s[%s]\n", (prev_section[0] ? "\n" : ""), section);
strncpy(prev_section, section, sizeof(prev_section));
prev_section[sizeof(prev_section) - 1] = '\0';
}
printf("%s = %s\n", name, value);
return 1;
}
int main(int argc, char* argv[])
{
int error;
if (argc <= 1) {
printf("Usage: ini_dump filename.ini\n");
return 1;
}
error = ini_parse(argv[1], dumper, NULL);
if (error < 0) {
printf("Can't read '%s'!\n", argv[1]);
return 2;
}
else if (error) {
printf("Bad config file (first error on line %d)!\n", error);
return 3;
}
return 0;
}

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/* Example: parse a simple configuration file */
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "../ini.h"
typedef struct
{
int version;
const char* name;
const char* email;
} configuration;
static int handler(void* user, const char* section, const char* name,
const char* value)
{
configuration* pconfig = (configuration*)user;
#define MATCH(s, n) strcmp(section, s) == 0 && strcmp(name, n) == 0
if (MATCH("protocol", "version")) {
pconfig->version = atoi(value);
} else if (MATCH("user", "name")) {
pconfig->name = strdup(value);
} else if (MATCH("user", "email")) {
pconfig->email = strdup(value);
} else {
return 0; /* unknown section/name, error */
}
return 1;
}
int main(int argc, char* argv[])
{
configuration config;
if (ini_parse("test.ini", handler, &config) < 0) {
printf("Can't load 'test.ini'\n");
return 1;
}
printf("Config loaded from 'test.ini': version=%d, name=%s, email=%s\n",
config.version, config.name, config.email);
return 0;
}

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/* Parse a configuration file into a struct using X-Macros */
#include <stdio.h>
#include <string.h>
#include "../ini.h"
/* define the config struct type */
typedef struct {
#define CFG(s, n, default) char *s##_##n;
#include "config.def"
} config;
/* create one and fill in its default values */
config Config = {
#define CFG(s, n, default) default,
#include "config.def"
};
/* process a line of the INI file, storing valid values into config struct */
int handler(void *user, const char *section, const char *name,
const char *value)
{
config *cfg = (config *)user;
if (0) ;
#define CFG(s, n, default) else if (strcmp(section, #s)==0 && \
strcmp(name, #n)==0) cfg->s##_##n = strdup(value);
#include "config.def"
return 1;
}
/* print all the variables in the config, one per line */
void dump_config(config *cfg)
{
#define CFG(s, n, default) printf("%s_%s = %s\n", #s, #n, cfg->s##_##n);
#include "config.def"
}
int main(int argc, char* argv[])
{
if (ini_parse("test.ini", handler, &Config) < 0)
printf("Can't load 'test.ini', using defaults\n");
dump_config(&Config);
return 0;
}

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; Test config file for ini_example.c and INIReaderTest.cpp
[protocol] ; Protocol configuration
version=6 ; IPv6
[user]
name = Bob Smith ; Spaces around '=' are stripped
email = bob@smith.com ; And comments (like this) ignored
active = true ; Test a boolean
pi = 3.14159 ; Test a floating point number

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# Simple makefile to build inih as a static library using g++
SRC = ../ini.c
OBJ = $(SRC:.c=.o)
OUT = libinih.a
INCLUDES = -I..
CCFLAGS = -g -O2
CC = g++
default: $(OUT)
.c.o:
$(CC) $(INCLUDES) $(CCFLAGS) $(EXTRACCFLAGS) -c $< -o $@
$(OUT): $(OBJ)
ar rcs $(OUT) $(OBJ) $(EXTRAARFLAGS)
clean:
rm -f $(OBJ) $(OUT)

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/* inih -- simple .INI file parser
inih is released under the New BSD license (see LICENSE.txt). Go to the project
home page for more info:
http://code.google.com/p/inih/
*/
#include <stdio.h>
#include <ctype.h>
#include <string.h>
#include "ini.h"
#if !INI_USE_STACK
#include <stdlib.h>
#endif
#define MAX_SECTION 50
#define MAX_NAME 50
/* Strip whitespace chars off end of given string, in place. Return s. */
static char* rstrip(char* s)
{
char* p = s + strlen(s);
while (p > s && isspace((unsigned char)(*--p)))
*p = '\0';
return s;
}
/* Return pointer to first non-whitespace char in given string. */
static char* lskip(const char* s)
{
while (*s && isspace((unsigned char)(*s)))
s++;
return (char*)s;
}
/* Return pointer to first char c or ';' comment in given string, or pointer to
null at end of string if neither found. ';' must be prefixed by a whitespace
character to register as a comment. */
static char* find_char_or_comment(const char* s, char c)
{
int was_whitespace = 0;
while (*s && *s != c && !(was_whitespace && *s == ';')) {
was_whitespace = isspace((unsigned char)(*s));
s++;
}
return (char*)s;
}
/* Version of strncpy that ensures dest (size bytes) is null-terminated. */
static char* strncpy0(char* dest, const char* src, size_t size)
{
strncpy(dest, src, size);
dest[size - 1] = '\0';
return dest;
}
/* See documentation in header file. */
int ini_parse_file(FILE* file,
int (*handler)(void*, const char*, const char*,
const char*),
void* user)
{
/* Uses a fair bit of stack (use heap instead if you need to) */
#if INI_USE_STACK
char line[INI_MAX_LINE];
#else
char* line;
#endif
char section[MAX_SECTION] = "";
char prev_name[MAX_NAME] = "";
char* start;
char* end;
char* name;
char* value;
int lineno = 0;
int error = 0;
#if !INI_USE_STACK
line = (char*)malloc(INI_MAX_LINE);
if (!line) {
return -2;
}
#endif
/* Scan through file line by line */
while (fgets(line, INI_MAX_LINE, file) != NULL) {
lineno++;
start = line;
#if INI_ALLOW_BOM
if (lineno == 1 && (unsigned char)start[0] == 0xEF &&
(unsigned char)start[1] == 0xBB &&
(unsigned char)start[2] == 0xBF) {
start += 3;
}
#endif
start = lskip(rstrip(start));
if (*start == ';' || *start == '#') {
/* Per Python ConfigParser, allow '#' comments at start of line */
}
#if INI_ALLOW_MULTILINE
else if (*prev_name && *start && start > line) {
/* Non-black line with leading whitespace, treat as continuation
of previous name's value (as per Python ConfigParser). */
if (!handler(user, section, prev_name, start) && !error)
error = lineno;
}
#endif
else if (*start == '[') {
/* A "[section]" line */
end = find_char_or_comment(start + 1, ']');
if (*end == ']') {
*end = '\0';
strncpy0(section, start + 1, sizeof(section));
*prev_name = '\0';
}
else if (!error) {
/* No ']' found on section line */
error = lineno;
}
}
else if (*start && *start != ';') {
/* Not a comment, must be a name[=:]value pair */
end = find_char_or_comment(start, '=');
if (*end != '=') {
end = find_char_or_comment(start, ':');
}
if (*end == '=' || *end == ':') {
*end = '\0';
name = rstrip(start);
value = lskip(end + 1);
end = find_char_or_comment(value, '\0');
if (*end == ';')
*end = '\0';
rstrip(value);
/* Valid name[=:]value pair found, call handler */
strncpy0(prev_name, name, sizeof(prev_name));
if (!handler(user, section, name, value) && !error)
error = lineno;
}
else if (!error) {
/* No '=' or ':' found on name[=:]value line */
error = lineno;
}
}
}
#if !INI_USE_STACK
free(line);
#endif
return error;
}
/* See documentation in header file. */
int ini_parse(const char* filename,
int (*handler)(void*, const char*, const char*, const char*),
void* user)
{
FILE* file;
int error;
file = fopen(filename, "r");
if (!file)
return -1;
error = ini_parse_file(file, handler, user);
fclose(file);
return error;
}

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/* inih -- simple .INI file parser
inih is released under the New BSD license (see LICENSE.txt). Go to the project
home page for more info:
http://code.google.com/p/inih/
*/
#ifndef __INI_H__
#define __INI_H__
/* Make this header file easier to include in C++ code */
#ifdef __cplusplus
extern "C" {
#endif
#include <stdio.h>
/* Parse given INI-style file. May have [section]s, name=value pairs
(whitespace stripped), and comments starting with ';' (semicolon). Section
is "" if name=value pair parsed before any section heading. name:value
pairs are also supported as a concession to Python's ConfigParser.
For each name=value pair parsed, call handler function with given user
pointer as well as section, name, and value (data only valid for duration
of handler call). Handler should return nonzero on success, zero on error.
Returns 0 on success, line number of first error on parse error (doesn't
stop on first error), -1 on file open error, or -2 on memory allocation
error (only when INI_USE_STACK is zero).
*/
int ini_parse(const char* filename,
int (*handler)(void* user, const char* section,
const char* name, const char* value),
void* user);
/* Same as ini_parse(), but takes a FILE* instead of filename. This doesn't
close the file when it's finished -- the caller must do that. */
int ini_parse_file(FILE* file,
int (*handler)(void* user, const char* section,
const char* name, const char* value),
void* user);
/* Nonzero to allow multi-line value parsing, in the style of Python's
ConfigParser. If allowed, ini_parse() will call the handler with the same
name for each subsequent line parsed. */
#ifndef INI_ALLOW_MULTILINE
#define INI_ALLOW_MULTILINE 1
#endif
/* Nonzero to allow a UTF-8 BOM sequence (0xEF 0xBB 0xBF) at the start of
the file. See http://code.google.com/p/inih/issues/detail?id=21 */
#ifndef INI_ALLOW_BOM
#define INI_ALLOW_BOM 1
#endif
/* Nonzero to use stack, zero to use heap (malloc/free). */
#ifndef INI_USE_STACK
#define INI_USE_STACK 1
#endif
/* Maximum line length for any line in INI file. */
#ifndef INI_MAX_LINE
#define INI_MAX_LINE 200
#endif
#ifdef __cplusplus
}
#endif
#endif /* __INI_H__ */

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[protocol]
version = 5 ; próba komment
[user]
name = Gipsz Jakab ; comment
email = gip@szja.kab ; comment

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#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <string.h>
#include "ini.h"
typedef struct {
int version;
const char* name;
const char* email;
} configuration;
static int handler(void* user, const char* section, const char* name,
const char* value)
{
configuration *pconfig = (configuration*)user;
#define MATCH(s, n) strcmp(section, s) == 0 && strcmp(name, n) == 0
if (MATCH("protocol", "version")) {
pconfig->version = atoi(value);
} else if (MATCH("user", "name")) {
pconfig->name = strdup(value);
} else if (MATCH("user", "email")) {
pconfig->email = strdup(value);
} else {
return 0; /* unknown section/name, error */
}
return 1;
}
int main (int argc, char* argv[]) {
configuration config;
if (ini_parse("test.ini", handler, &config) < 0) {
printf("Can't load 'test.ini'\n");
return 1;
}
printf("Config loaded from 'test.ini': version=%d, name=%s, email=%s\n",
config.version, config.name, config.email);
return 0;
}

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This is an error

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indented

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[section1]
name1=value1
[section2
[section3 ; comment ]
name2=value2

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no_file.ini: e=-1 user=0
... [section1]
... one=This is a test;
... two=1234;
... [ section 2 ]
... happy=4;
... sad=;
... [comment_test]
... test1=1;2;3;
... test2=2;3;4;this won't be a comment, needs whitespace before ';';
... test;3=345;
... test4=4#5#6;
... [colon_tests]
... Content-Type=text/html;
... foo=bar;
... adams=42;
normal.ini: e=0 user=101
... [section1]
... name1=value1;
... name2=value2;
bad_section.ini: e=3 user=102
bad_comment.ini: e=1 user=102
... [section]
... a=b;
... user=parse_error;
... c=d;
user_error.ini: e=3 user=104
... [section1]
... single1=abc;
... multi=this is a;
... multi=multi-line value;
... single2=xyz;
... [section2]
... multi=a;
... multi=b;
... multi=c;
... [section3]
... single=ghi;
... multi=the quick;
... multi=brown fox;
... name=bob smith;
multi_line.ini: e=0 user=105
bad_multi.ini: e=1 user=105
... [bom_section]
... bom_name=bom_value;
... key“=value“;
bom.ini: e=0 user=107

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no_file.ini: e=-1 user=0
... [section1]
... one=This is a test;
... two=1234;
... [ section 2 ]
... happy=4;
... sad=;
... [comment_test]
... test1=1;2;3;
... test2=2;3;4;this won't be a comment, needs whitespace before ';';
... test;3=345;
... test4=4#5#6;
... [colon_tests]
... Content-Type=text/html;
... foo=bar;
... adams=42;
normal.ini: e=0 user=101
... [section1]
... name1=value1;
... name2=value2;
bad_section.ini: e=3 user=102
bad_comment.ini: e=1 user=102
... [section]
... a=b;
... user=parse_error;
... c=d;
user_error.ini: e=3 user=104
... [section1]
... single1=abc;
... multi=this is a;
... single2=xyz;
... [section2]
... multi=a;
... [section3]
... single=ghi;
... multi=the quick;
... name=bob smith;
multi_line.ini: e=4 user=105
bad_multi.ini: e=1 user=105
... [bom_section]
... bom_name=bom_value;
... key“=value“;
bom.ini: e=0 user=107

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[bom_section]
bom_name=bom_value
key“ = value“

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[section1]
single1 = abc
multi = this is a
multi-line value
single2 = xyz
[section2]
multi = a
b
c
[section3]
single: ghi
multi: the quick
brown fox
name = bob smith ; comment line 1
; comment line 2

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; This is an INI file
[section1] ; section comment
one=This is a test ; name=value comment
two = 1234
; x=y
[ section 2 ]
happy = 4
sad =
[empty]
; do nothing
[comment_test]
test1 = 1;2;3 ; only this will be a comment
test2 = 2;3;4;this won't be a comment, needs whitespace before ';'
test;3 = 345 ; key should be "test;3"
test4 = 4#5#6 ; '#' only starts a comment at start of line
#test5 = 567 ; entire line commented
# test6 = 678 ; entire line commented, except in MULTILINE mode
[colon_tests]
Content-Type: text/html
foo:bar
adams : 42

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@call tcc ..\ini.c -I..\ -run unittest.c > baseline_multi.txt
@call tcc ..\ini.c -I..\ -DINI_ALLOW_MULTILINE=0 -run unittest.c > baseline_single.txt

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/* inih -- unit tests
This works simply by dumping a bunch of info to standard output, which is
redirected to an output file (baseline_*.txt) and checked into the Subversion
repository. This baseline file is the test output, so the idea is to check it
once, and if it changes -- look at the diff and see which tests failed.
Here's how I produced the two baseline files (with Tiny C Compiler):
tcc -DINI_ALLOW_MULTILINE=1 ../ini.c -run unittest.c > baseline_multi.txt
tcc -DINI_ALLOW_MULTILINE=0 ../ini.c -run unittest.c > baseline_single.txt
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "../ini.h"
int User;
char Prev_section[50];
int dumper(void* user, const char* section, const char* name,
const char* value)
{
User = (int)user;
if (strcmp(section, Prev_section)) {
printf("... [%s]\n", section);
strncpy(Prev_section, section, sizeof(Prev_section));
Prev_section[sizeof(Prev_section) - 1] = '\0';
}
printf("... %s=%s;\n", name, value);
return strcmp(name, "user")==0 && strcmp(value, "parse_error")==0 ? 0 : 1;
}
void parse(const char* fname) {
static int u = 100;
int e;
*Prev_section = '\0';
e = ini_parse(fname, dumper, (void*)u);
printf("%s: e=%d user=%d\n", fname, e, User);
u++;
}
int main(void)
{
parse("no_file.ini");
parse("normal.ini");
parse("bad_section.ini");
parse("bad_comment.ini");
parse("user_error.ini");
parse("multi_line.ini");
parse("bad_multi.ini");
parse("bom.ini");
return 0;
}

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[section]
a = b
user = parse_error
c = d

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/* logger.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <time.h>
#include <sys/time.h>
#include <stdarg.h>
#include "logger.h"
#include "debug.h"
#include "misc.h"
#include "utlist.h"
#include "misc.h"
//~ #define concat(dst, src) strncat((dst), (src), ((sizeof(dst)) - strlen(dst) - 1))
//~ #define concatf(dst, ...) snprintf((dst) + strlen((dst)), sizeof((dst)) - strlen((dst)), __VA_ARGS__)
static char *logfile = NULL;
static char logfile_buf[128];
/* TODO: rendbe tenni ezt a marhaságot valami kultúrált megoldással úgy,
hogy a hívó függvény által átadott stringet lemásoljuk és helyben a
másolatot kezeljük. Azért, hogy a hívó utána a saját mutatójával azt
csinálhasson, amit csak akar. Kezelni a NULL-t, a default értékeket és
valami kultúrált köntöst adni neki, ne ezt a három változós izét. */
static char *timestamp_format_default = "%a %H:%M:%S";
static char *timestamp_format = "%a %H:%M:%S";
static char timestamp_format_buf[128];
static int initialised = 0;
struct timeval first_time = {0, 0};
char ela[16];
typedef struct el {
struct timeval tm;
char *line;
struct el *prev; /* needed for a doubly-linked list only */
struct el *next; /* needed for singly- or doubly-linked lists */
} el;
/* Ha a tv (első paraméter) meg van adva, akkor nincs gettimeofday hívás, hanem
* a tv-ben beállított idő lesz a mérvadó. Ha a tv értéke NULL, akkor a
* jelenlegi idővel (gettimeofday) számolunk. Ha meg van adva a ptm, akkor az is
* be lesz állítva. Ha az elapsed nem NULL, akkor kiszámolja az indítástól
* eltelt időt, és visszaad egy mutatót, ami a kiszámolt érték ssss.mmmu
* formátumú stringre mutat.
*/
static struct timeval get_time (struct timeval *tv, struct tm **ptm, char *elapsed) {
struct timeval mytv;
if (tv == NULL) {
gettimeofday(&mytv, NULL);
} else {
mytv.tv_sec = tv->tv_sec;
mytv.tv_usec = tv->tv_usec;
}
if (tv)
*tv = mytv;
if (ptm)
*ptm = localtime(&mytv.tv_sec); // thread safe?
if (elapsed) {
long int delta = ((mytv.tv_sec * 1000000) + mytv.tv_usec) - ((first_time.tv_sec * 1000000) + first_time.tv_usec);
snprintf(elapsed, 8, "%3.3f", (float)delta / 1000000);
}
return mytv;
}
void con_init () {
if (initialised)
return;
first_time = get_time(NULL, NULL, NULL);
initialised = 1;
return;
}
void con_logfile (const char *file) {
if (file == NULL) {
logfile = NULL;
return;
}
strncpy(logfile_buf, file, sizeof(logfile_buf));
logfile = logfile_buf;
}
void con_timestamp_format (const char *format) {
if (format == NULL) {
timestamp_format = timestamp_format_default;
return;
}
strncpy(timestamp_format_buf, format, sizeof(timestamp_format_buf));
timestamp_format = timestamp_format_buf;
}
// gány, memóriazabáló függvény :)
void _con_writef (enum con_callmode cm, char *file, int line, const char *function, const char *fmt, ...) {
if (!initialised)
con_init();
// botrányos deklarációk
char tmp[1024] = {0}; // ide kerül a végeredmény
char tmp2[1024]; // va_arg szöveges kimenete
char timestamp[64]; // timestamp
char elapsed[64]; // elapsed
va_list ap;
va_start(ap, fmt);
vsnprintf(tmp2, sizeof(tmp2), fmt, ap);
va_end(ap);
// TODO: ezt az idő lekérdezős részt külön függvénybe kell rakni!
struct timeval tv;
struct tm *ptm;
tv = get_time(NULL, &ptm, NULL); // idő, ami alapján a továbbiakban számolunk
strftime(timestamp, sizeof(timestamp), timestamp_format, ptm); // szöveges timestamp
get_time(&tv, &ptm, elapsed); // elapsed lekérdezése TODO: ez borzalmas, átírni!!!
switch (cm) {
case CON_CALLMODE_CONFT:
case CON_CALLMODE_CONFTN:
concatf(tmp, "%s %s", timestamp, tmp2);
break;
case CON_CALLMODE_CONF:
case CON_CALLMODE_CONFN:
strncpy(tmp, tmp2, sizeof(tmp) - 1);
break;
case CON_CALLMODE_DEBUG:
concatf(tmp, "%s /%s/ [%s:%d %s]: %s", timestamp, elapsed, file, line, function, tmp2);
break;
}
chomp(tmp);
// Kiiratás
printf("%s", tmp);
if (cm != CON_CALLMODE_CONFTN && cm != CON_CALLMODE_CONFN)
printf("\n");
fflush(stdout);
if (logfile) {
FILE *f = fopen(logfile, "a");
if (f != NULL) {
fprintf(f, "%s", tmp);
if (cm != CON_CALLMODE_CONFTN && cm != CON_CALLMODE_CONFN)
fprintf(f, "\n");
fclose(f);
}
}
}

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/* logger.h
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*
conf(...) kiírás időbélyeg nélkül, újsorral
confn(...) kiírás időbélyeg nélkül, újsor nélkül
conft(...) kiírás időbélyeggel, újsorral
conftn(...) kiírás időbélyeggel, újsor nélkül
*/
enum con_callmode {
CON_CALLMODE_CONF,
CON_CALLMODE_CONFN,
CON_CALLMODE_CONFT,
CON_CALLMODE_CONFTN,
CON_CALLMODE_DEBUG,
};
#ifndef LOGGER_H_LOADED
#define LOGGER_H_LOADED
void con_init ();
//~ void conft (const char *fmt, ...);
void con_logfile (const char *file);
void con_timestamp_format (const char *format);
void _con_writef (enum con_callmode cm, char *file, int line, const char *function, const char *fmt, ...);
#endif
// ha a DEBUG makró 1, akkor a debug() makrók életbe lépnek,
// ellenkező esetben a kódba sem kerül bele :)
#ifdef CON_DEBUG
#define con_debug(...) _con_writef(CON_CALLMODE_DEBUG, __FILE__, __LINE__, __FUNCTION__, __VA_ARGS__)
#else
#define con_debug(...)
#endif
#define conf(...) _con_writef(CON_CALLMODE_CONF, NULL, 0, NULL, __VA_ARGS__)
#define confn(...) _con_writef(CON_CALLMODE_CONFN, NULL, 0, NULL, __VA_ARGS__)
#define conft(...) _con_writef(CON_CALLMODE_CONFT, NULL, 0, NULL, __VA_ARGS__)
#define conftn(...) _con_writef(CON_CALLMODE_CONFTN, NULL, 0, NULL, __VA_ARGS__)

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/* logger_test.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#define CON_DEBUG
#include "logger.h"
#include <stdio.h>
#include <errno.h>
int main () {
con_logfile("logger_test.log");
con_timestamp_format("%a %H:%M:%S");
conft("zsiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiiir");
conft("yeahhhh");
conft("ok");
FILE *f;
if (!(f = fopen("/proc/ioports", "r"))) {
perror("fopen");
return 1;
}
char tmp[128];
int i = 0;
while (fgets(tmp, sizeof(tmp), f)) {
con_debug("i=%d %s", i, tmp);
i++;
}
fclose(f);
return 0;
}

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/* misc.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/* altalanos fuggvenyek */
#include <stdlib.h>
#include <string.h>
#include <stdio.h>
#include <arpa/inet.h> // inet_pton() függvény
#include <stdarg.h>
#include "debug.h"
char *chomp (char *s) {
if (s == NULL)
return NULL;
size_t len = strlen(s);
int n;
for (n = len - 1; n >= 0; n--) {
if (s[n] == '\n' || s[n] == '\r')
s[n] = '\0';
else
return s;
}
return s;
}
/*
fmtsub - mintából, változó-érték párokból és készít stringet behelyettesítéssel
paraméterek:
dest ebbe a stringbe menti el az eredményt
size dest mérete
pattern minta
args változó-érték táblázat, nullával lezárva
visszatérési érték:
0 sikerült a behelyettesítés
-1 valamelyik mutató == NULL vagy a size == 0
A táblázat megadása:
const char *args[][2] = {
{"%n", "2463434"},
{"%v", "0.9.6-rc2"},
{"version", "Linux 2.6.32"},
{"boo", "baa"},
{0}}; */
int fmtsub (char *dest, size_t size, const char *pattern, const char *args[][2]) {
char buf[4096];
//~ debs(dest); debi(size); debs(pattern); debi(args);
if (dest == NULL || pattern == NULL || args == NULL || size == 0)
return -1;
// ha a pattern üres string, akkor üres stringet adunk vissza
if (strlen(pattern) == 0) {
strcpy(dest, "");
return 0;
}
int pospat, posbuf, a;
int lenpat = strlen(pattern);
buf[0] = '\0';
// byteonként végigmegyünk a pattern stringen
for (pospat = 0, posbuf = 0; pospat < lenpat && posbuf < sizeof(buf) - 1;) {
const char *subpat = &pattern[pospat];
int match = 0;
// végigmegyünk az args táblázaton
for (a = 0; args[a][0] != 0; a++) {
const char *var = args[a][0]; // "%n"
const char *exp = args[a][1]; // "2463434"
// ha a változónév megtalálható a pattern jelenlegi pozíciójánál
if (strlen(var) != 0 && !strncmp(subpat, var, strlen(var))) {
if (exp != NULL) {
// bemásoljuk a változó értékét a bufferbe
strncat(buf, exp, sizeof(buf) - posbuf - 1);
posbuf += strlen(exp);
}
/* ezen a ponton a posbuf nagyobb lehet, mint a buf, ami
igen kellemetlenül érintené a lenti buf[posbuf] = '\0'
értékadást, ezért a posbuf pozícionálót beállítjuk a buf
utolsó byte-jára. ide fog kerülni a NULL. */
if (posbuf >= sizeof(buf) - 1)
posbuf = sizeof(buf) - 1;
// pattern pozícióját a változónév utáni karakterre állítjuk
pospat += strlen(var);
// kihagyjuk a lenti if-et
match = 1;
break;
}
}
// 1 bájt másolása pattern-ből buf-ba
if (!match) {
buf[posbuf] = pattern[pospat];
buf[posbuf+1] = '\0';
pospat++;
posbuf++;
}
}
buf[posbuf] = '\0'; // lezáró NULL
strncpy(dest, buf, size - 1);
return 0;
}
// http://en.wikipedia.org/wiki/ROT13
// paraméter az a string, amit módosítani kell
// visszatérés ugyan az, mint a paraméter kompatibilitási izé miatt
char *encode_rot13 (char *s) {
if (s == NULL)
return NULL;
int i;
for (i = 0; s[i]; i++) {
if (s[i] >= 'a' && s[i] <= 'm') { s[i] += 13; continue; }
if (s[i] >= 'A' && s[i] <= 'M') { s[i] += 13; continue; }
if (s[i] >= 'n' && s[i] <= 'z') { s[i] -= 13; continue; }
if (s[i] >= 'N' && s[i] <= 'Z') { s[i] -= 13; continue; }
}
return s;
}
// szóközöket és tabokat vág le a megadott string bal és/vagy jobb oldaláról
// trim - mindkét oldalról vág
// ltrim - bal oldalról vág
// rtrim - jobb oldalról vág
char *_trim (char *s, int trim_from_left, int trim_from_right) {
if (s == NULL)
return NULL;
int n, i;
size_t len = strlen(s);
if (trim_from_right) {
for (n = len - 1; n >= 0; n--) {
if (s[n] == 32 || s[n] == '\t') {
s[n] = '\0';
len--;
} else {
break;
}
}
}
if (trim_from_left) {
for (n = 0; n < len; n++)
if (s[n] != 32 && s[n] != '\t')
break;
if (n > 0) {
for (i = n; i < len; i++)
s[i-n] = s[i];
for (i = len - n; i < len; i++)
s[i] = '\0';
}
}
return s;
}
/* megszámlálja a <delimeter> elemeket */
static int split_get_size (char *buffer, int delimeter) {
int c = 0, i = 0;
if (buffer[ 0 ] == 0)
return 0;
while (buffer[i] != 0) {
if (buffer[i] == delimeter)
c++;
i++;
}
/* egyet hozzáadok, mert a "hello" is értéknek számít, hiába nincs benne ';' */
return c + 1;
}
/* visszatér egy pointer tömbbel, ami az elemekre mutat, a tömböt 0 pointerrel zárja */
char **split (char *buffer, int delimeter) {
int size = split_get_size(buffer, delimeter);
/* helyfoglalás a pointer tömbnek */
char **res = (char **)malloc(sizeof(char *) * (size + 1));
int i=0;
int p=0;
while (buffer[i] != 0) {
/* a sor elejét eltárolom */
res[p++] = buffer + i;
/* a sor végét megkeresem */
do {
i++;
} while (buffer[i] != 0 && buffer[i] != delimeter);
/* felülírjuk a ';'-t 0-val */
if (buffer[i] != 0) {
buffer[i] = 0;
i++;
}
}
/* null pointerrel zárjuk */
res[p] = 0;
return res;
}
// 1-et ad vissza, ha érvényes a kapott IP cím
// 0-át ha nem
int is_valid_ip (const char *ip) {
struct sockaddr_in sa;
int result = inet_pton(AF_INET, ip, &(sa.sin_addr));
return result != 0;
}
void die (const char *fmt, ...) {
va_list ap;
va_start(ap, fmt);
vfprintf(stderr, fmt, ap);
va_end(ap);
exit(-1);
}
void read_lines_from_stdin (char *dst, int size) {
char *line = NULL; // ideiglenes buffer a getline-nak
size_t len = 0;
ssize_t read;
int remaining_size; // dst stringbe még ennyi byte fér el
remaining_size = size - 1;
while ((read = getline(&line, &len, stdin)) != -1) {
strncat(dst, line, remaining_size);
remaining_size -= read;
if (remaining_size <= 0)
break;
}
chomp(dst); // utolsó \n karakter chompolása
free(line);
}
char *strcutpbrk (char *str, const char *accept) {
if (str == NULL)
return NULL;
if (accept == NULL)
return str;
char *at = strpbrk(str, accept);
if (at != NULL)
*at = '\0';
return str;
}
char *strdelchars (char *str, const char *dels) {
if (!str)
return NULL;
if (!dels)
return str;
int len_str = strlen(str);
int len_dels = strlen(dels);
if (!len_str || !len_dels)
return str;
int i, j, k = 0;
for (i = 0; i < len_str; i++) {
for (j = 0; j < len_dels; j++)
if (str[i] == dels[j])
break;
if (j == len_dels) // ha nem volt break
str[k++] = str[i];
}
str[k] = '\0';
return str;
}

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/* misc.h
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
char *chomp (char *str);
#define scpy(dst, src) strncpy((dst), (src), (sizeof(dst)) - 1)
#define concat(dst, src) strncat((dst), (src), ((sizeof(dst)) - strlen(dst) - 1))
#define concatf(dst, ...) snprintf((dst) + strlen((dst)), sizeof((dst)) - strlen((dst)), __VA_ARGS__)
int fmtsub (char *dest, size_t size, const char *pattern, const char *args[][2]);
char *encode_rot13 (char *s);
char *_trim (char *s, int trim_from_left, int trim_from_right);
#define trim(s) _trim(s, 1, 1)
#define ltrim(s) _trim(s, 1, 0)
#define rtrim(s) _trim(s, 0, 1)
// Az str stringet levágja ott, ahol az accept stringben felsorolt első
// karakterrel találkozik. A levágást úgy oldja meg, hogy az str string
// megfelelő byte-ját átírja 0-ra. A visszatérési érték az str string. Ha az
// accept == NULL, akkor nem piszkálja az str stringet. Ha az str == NULL,
// akkor a visszatérési érték NULL. Például ha az str "Bandi" és az accept
// "cde", akkor az eredmény "Ban" lesz.
char *strcutpbrk (char *str, const char *accept);
// Nincs még doksi, ezért ide írom...
// A chomp, trim, encode_rot13, strcutpbrk függvények használhatok beágyazva is, tehát:
// char buf[64];
// encode_rot13(trim(chomp(buf)));
// visszatér egy pointer tömbbel, ami az elemekre mutat, a tömböt 0 pointerrel zárja. A függvény felülírja a buffer stringben lévő delimetereket. Magyarul módosítja a kapott stringet, ami nem biztos, hogy jó dolog. Pl. char *x = "hello" változóknál segfault.
char **split (char *buffer, int delimeter);
// 1-et ad vissza, ha érvényes a kapott IP cím
// 0-át ha nem
int is_valid_ip (const char *ip);
// mint Perl-ben:)
void die (const char *fmt, ...);
// STDIN-ről beolvasás EOF-ig. A sorokat összefűzi és a dst stringbe menti,
// vigyázva a lezáró NULL karakterre és a dst string hosszára, amit a size-ben
// kell átadni. Az utolsó \n karakterek le lesznek csípve.
void read_lines_from_stdin (char *dst, int size);
// Az str stringből eltávolítja a dels stringben megtalálható karaktereket. Az
// eltávolítás folyamán az str stringben a byteokat fokozatosan balra rendezi.
// A művelet végén elhelyezi a stringet lezáró NULL karaktert. Az eredmény
// ugyan akkora vagy kisebb méretű string. A visszatérési érték az str string.
// Ha az str NULL, akkor a visszatérési érték NULL. Ha a dels NULL, akkor a
// visszatérési érték az str string.
char *strdelchars (char *str, const char *dels);

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/* netsocket.c
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
// TODO: datapipe-nál jól látható, hogy a connect előtt bejött adatok
// nem kerülnek kiküldésre. Valami buffer megoldást kéne alkalmazni. vagy mégsem?
// ... végülis a Centauri nézőpontja szerint connect előtt miért küldenénk adatokat?
// TODO: gethostbyaddr és gethostbyname függvényeket lecserélni getaddrinfo és
// getnameinfo függvényekre
//
// Figyelem! Amelyik függvényben invoke_callback van, annak a legaljára kell
// destroy_netsocket. Na meg ezt a kommentet átfogalmazni, mert valószínűleg
// baromira nem fogom érteni két hét múlva, hogy mi a szar is akar ez lenni
#include <stdio.h>
#include <errno.h>
#include <unistd.h>
#include <malloc.h>
#include <string.h>
#include <sys/socket.h>
#include <resolv.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <fcntl.h>
#include <stdarg.h>
#include "netsocket.h"
#include "debug.h"
static void dummy_callback (netsocket_t *obj, int event) {
printf("dummy_callback called at %s:%d\n", __FILE__, __LINE__);
printf(" event = %d\n mode = %s\n host = %s\n port = %d\n lhost = %s\n lport = %d\n",
event,
(obj->mode == NETSOCKET_SERVER) ? "server" : "client",
obj->host,
obj->port,
obj->lhost,
obj->lport
);
}
static void sock_close (netsocket_t *obj) {
if (!obj->sock)
return;
ev_io_stop(obj->loop, &obj->w_in);
ev_io_stop(obj->loop, &obj->w_out);
ev_timer_stop(obj->loop, &obj->w_connect_timeout);
close(obj->sock);
obj->sock = 0;
}
static void invoke_callback (netsocket_t *obj, int event) {
obj->in_callback++;
obj->callback(obj, event);
obj->in_callback--;
// késleltetett destroy
if (obj->destroy_request) {
if (!obj->in_callback) {
netsocket_destroy(obj);
}
}
}
static void disconnect (netsocket_t *obj, char *reason, int ignore_callback) {
if (obj == NULL)
return;
sock_close(obj);
if (reason != NULL)
strncpy(obj->disconnect_reason, reason, sizeof(obj->disconnect_reason) - 1);
if (!ignore_callback)
invoke_callback(obj, NETSOCKET_EVENT_DISCONNECT);
//~ if (!obj->connected)
//~ return;
obj->connected = 0;
// automatikusan felszabaditjuk a gyermek objektumot abban az esetben, ha
// az obj kliens modban van es bejovo kapcsolatrol van szo, maskeppen mondva
// akkor, ha egy szerver objektum gyermek objektuma szakad meg
// ilyen objektumot a sock_accept allokal malloccal
//~ if (obj->mode == NETSOCKET_CLIENT && obj->direction == NETSOCKET_IN)
//~ netsocket_destroy(obj);
}
/**
* @brief Disconnect from peer, close socket
* @param obj the netsocket object
* @param reason description of disconnect reason
*
* If connection in progress, abort the connection and close the socket.
* If connection was established, disconnect from peer, close socket and
* detach from event loop. Call the callback with NETSOCKET_EVENT_DISCONNECT
* event. Save the "reason" to obj->reason.
*/
void netsocket_disconnect (netsocket_t *obj, char *reason) {
disconnect(obj, reason, 1); // 1 jelzi, hogy nem kérünk callback hívást
}
void netsocket_disconnect_withevent (netsocket_t *obj, char *reason) {
disconnect(obj, reason, 0); // 0 jelzi, hogy kérünk callback hívást
}
// ha ezt meghívjuk, akkor a bejövő kapcsolatoknál nem kérdezzük meg
// a kliens IP címének nevét a DNS szervertől
void netsocket_disable_lookup_on_accept (netsocket_t *obj) {
obj->disable_lookup_on_accept = 1;
}
static int sock_accept (netsocket_t *parent) {
netsocket_t *obj = netsocket_new(dummy_callback, NULL, parent->loop);
obj->mode = NETSOCKET_CLIENT;
obj->direction = NETSOCKET_IN;
socklen_t addrlen = sizeof(obj->addr);
obj->sock = accept(parent->sock, (struct sockaddr *) &obj->addr, &addrlen);
if (obj->sock < 0) { // TODO rendes hibakezelés, memória felszabadítás
perror("accept"); // például: Too many open files
netsocket_destroy(obj);
return -1;
}
// kliens IP cím lekérdezése, ha nincs beállítva a "disable_lookup_on_accept"
if (!parent->disable_lookup_on_accept)
obj->hostent = gethostbyaddr(&obj->addr.sin_addr, sizeof(&obj->addr.sin_addr), AF_INET);
// ha nincs hostja az IP-nek, akkor a hostent NULL lesz
if (obj->hostent != NULL)
strncpy(obj->host, obj->hostent->h_name, sizeof(obj->host) - 1);
// IP cím tárolása szöveges formátumban
strncpy(obj->ip, inet_ntoa(obj->addr.sin_addr), sizeof(obj->ip) - 1);
// ha nincs hostja az IP-nek, akkor az IP lesz a host
if (!strlen(obj->host))
strncpy(obj->host, obj->ip, sizeof(obj->host) - 1);
// Kliens portja
obj->port = ntohs(obj->addr.sin_port);
ev_io_set(&obj->w_in, obj->sock, EV_READ);
ev_io_set(&obj->w_out, obj->sock, EV_WRITE);
ev_io_start(obj->loop, &obj->w_in);
ev_io_start(obj->loop, &obj->w_out);
obj->parent = parent;
obj->callback = parent->callback;
obj->userdata = parent->userdata;
strncpy(obj->lhost, parent->lhost, sizeof(obj->lhost));
obj->lport = parent->lport;
return 0;
}
static void w_connect_timeout_cb (EV_P_ ev_io *w, int revents) {
netsocket_t *obj = w->data;
disconnect(obj, "Connection timed out", 0);
}
static void w_in_cb (EV_P_ ev_io *w, int revents) {
netsocket_t *obj = w->data;
int i;
// ha a szerver portra csatlakozott új kliens
if (obj->mode == NETSOCKET_SERVER) {
sock_accept(obj);
return;
}
/* Ide akkor kerülünk, amikor egy kliensről adat érkezik.
* A NETSOCKET_EVENT_CONNECT eseményt és az obj->connected 1-re állítását
* a w_out_cb() függvény okozza, de az valójában a sock_accept() függvényben
* lenne esedékes. Szervernél, amikor várjuk a klienseket, normális esetben
* nincs ezzel semmmi gond, de ha valgrind-el fut, akkor valamiért
* összekeveredik az event loop-ban a sorrend és előbb hívódik meg az
* új kliens objektummal a NETSOCKET_EVENT_READ, mint a NETSOCKET_EVENT_CONNECT.
* Ez a hívó kódjában okozhat kollóziót (és okozott is), ezért ha nincs
* beállítva az obj->connected, de mégis adatot akarnánk beolvasni, akkor
* egyszerűen csak "elnapoljuk" a feladatot. Ekkor az event loop megteszi
* prevenciós körét a w_out_cb() függvényben is, ahol megtörténik a
* NETSOCKET_EVENT_CONNECT és utána ismét visszatér ide. Ezzel kényszerítjük
* ki a helyes sorrendet.
*/
if (!obj->connected)
return;
// adat beolvasása a kliensről
i = read(obj->sock, obj->inbuf, sizeof(obj->inbuf));
if (i < 1) {
disconnect(obj, (i == 0) ? "Connection reset by peer" : strerror(errno), 0);
} else {
obj->inbuf_len = i;
invoke_callback(obj, NETSOCKET_EVENT_READ);
}
}
static int sockerr (int sock) {
int optval = 0;
int err;
socklen_t optlen = sizeof(optval);
err = getsockopt(sock, SOL_SOCKET, SO_ERROR, &optval, &optlen);
if (err) return err;
return optval;
}
static void w_out_cb (EV_P_ ev_io *w, int revents) {
netsocket_t *obj = w->data;
//~ printf("Can't connect to %s: %s\n", obj->host, strerror(sockerr(obj->sock)));
obj->err = sockerr(obj->sock);
ev_io_stop(EV_A_ w);
if (obj->err) {
disconnect(obj, strerror(obj->err), 0);
} else {
obj->connected = 1;
ev_timer_stop(obj->loop, &obj->w_connect_timeout);
invoke_callback(obj, NETSOCKET_EVENT_CONNECT);
}
}
/**
* @brief Make outgoing connection to remote host
* @param obj the netsocket object
* @returns 0 if everything ok, otherwise -1
*
* Use netsocket as client. The function check the syntax of the "host" and "port" variable and
* lookup the host. If the host resolved successfully, then create an
* non-blocking outgoing socket, call connect(3) and add the socket to
* the event loop. In case of problem, close socket and call callback
* with event NETSOCKET_EVENT_DISCONNECT.
*/
int netsocket_connect (netsocket_t *obj) {
obj->mode = NETSOCKET_CLIENT;
obj->direction = NETSOCKET_OUT;
if (obj->host == NULL) {
disconnect(obj, "Invalid host", 0);
return -1;
}
if (obj->port < 1 || obj->port > 65535) {
disconnect(obj, "Invalid port", 0);
return -1;
}
// TODO aszinkronná tenni a host lookup-ot
if ((obj->hostent = gethostbyname(obj->host)) == NULL) {
disconnect(obj, "Host not found", 0);
return -1;
}
struct in_addr **pptr;
pptr = (struct in_addr **)obj->hostent->h_addr_list;
strncpy(obj->ip, inet_ntoa(**(pptr++)), sizeof(obj->ip));
obj->sock = socket(PF_INET, SOCK_STREAM, 0);
ev_io_set(&obj->w_in, obj->sock, EV_READ);
ev_io_set(&obj->w_out, obj->sock, EV_WRITE);
ev_timer_set(&obj->w_connect_timeout, (float)obj->connect_timeout / 1000, 0);
ev_io_start(obj->loop, &obj->w_in);
ev_io_start(obj->loop, &obj->w_out);
ev_timer_start(obj->loop, &obj->w_connect_timeout);
// socket non-block
fcntl(obj->sock, F_SETFL, fcntl(obj->sock, F_GETFL, 0) | O_NONBLOCK);
bzero(&obj->addr, sizeof(obj->addr));
obj->addr.sin_family = AF_INET;
obj->addr.sin_port = htons(obj->port);
obj->addr.sin_addr.s_addr = *(long*)(obj->hostent->h_addr);
connect(obj->sock, (struct sockaddr*)&obj->addr, sizeof(obj->addr));
if (errno != EINPROGRESS) {
netsocket_disconnect(obj, strerror(errno));
return -1;
}
return 0;
}
/**
* @brief Listening for incoming connections
* @param obj the netsocket object
* @returns 0 if everything ok, otherwise -1
*
* Use netsocket as server. The function check the syntax of
* the "lhost" and "lport" variable and
* lookup the lhost. If the host resolved successfully, then create an
* non-blocking incoming socket, bind to lhost, and add the socket to
* the event loop. If "lhost" is NULL, then "0.0.0.0" is assumed.
*
* When a peer connects to the listening port, create a new netsocket object and inherit
* lhost, lport, userdata and callback variables from the server object.
* Accept the incoming connection,
* add them to the event loop and call the callback with NETSOCKET_EVENT_CONNECT event.
* The callback's object parameter is the newly created client object. The server object's
* address is in obj->parent pointer. For each incoming connection make individually a
* new netsocket object.
*/
int netsocket_listen (netsocket_t *obj) {
obj->mode = NETSOCKET_SERVER;
if (!strlen(obj->lhost))
strcpy(obj->lhost, "0.0.0.0");
if (obj->lport < 1 || obj->lport > 65535) {
disconnect(obj, "Invalid local port", 0);
return -1;
}
// TODO aszinkronná tenni a host lookup-ot
if ((obj->hostent = gethostbyname(obj->lhost)) == NULL) {
disconnect(obj, "Local host not found", 0);
return -1;
}
struct in_addr **pptr;
pptr = (struct in_addr **)obj->hostent->h_addr_list;
strncpy(obj->ip, inet_ntoa(**(pptr++)), sizeof(obj->ip));
// TODO: socket függvény hibájának csekkolása
obj->sock = socket(AF_INET, SOCK_STREAM, 0);
fcntl(obj->sock, F_SETFL, fcntl(obj->sock, F_GETFL, 0) | O_NONBLOCK);
int optval = 1;
if (setsockopt(obj->sock, SOL_SOCKET, SO_REUSEADDR, &optval, sizeof(optval)) == -1) {
perror("setsockopt"); // TODO normális hibakezelés
return -1;
}
bzero(&obj->addr, sizeof(obj->addr));
obj->addr.sin_family = AF_INET;
obj->addr.sin_port = htons(obj->lport);
obj->addr.sin_addr.s_addr = *(long*)(obj->hostent->h_addr);
if (bind(obj->sock, (struct sockaddr *) &obj->addr, sizeof(obj->addr)) < 0) {
disconnect(obj, "Error on bind()", 0); // TODO rendes hibaüzenet
return -1;
}
if (listen(obj->sock, 5)) {
disconnect(obj, "Error on listen()", 0);
return -1;
}
ev_io_set(&obj->w_in, obj->sock, EV_READ);
ev_io_set(&obj->w_out, obj->sock, EV_WRITE);
//~ ev_timer_set(&obj->w_connect_timeout, (float)obj->connect_timeout / 1000, 0);
ev_io_start(obj->loop, &obj->w_in);
ev_io_start(obj->loop, &obj->w_out);
//~ ev_timer_start(EV_DEFAULT, &obj->w_connect_timeout);
return 0;
}
/**
* @brief write data to netsocket socket
* @param obj the netsocket object
* @param data data string
* @param length length of data string
* @returns length of written data
*
* Write data to socket. O yeahhh:)
*/
int netsocket_write (netsocket_t *obj, const char *data, int length) {
int i = 0;
if (!obj->sock) return 0; // véletlenül sem írunk az stdout-ra :)
/* Úgy néz ki, hogy Linux alatt a send() az igazi. Az MSG_NOSIGNAL nélkül a
rendszer PIPE szignállal kinyírja a szervert abban az esetben, ha "Broken
pipe" állapot lép fel, tehát a socket bezárult (kliens megszakadt) de a
szerver még írni akar . Az MSG_NOSIGNAL állítólag csak Linux alatt
létezik. BSD alatt pl. az a megoldás, hogy a setsockopt függvénnyel be kell
állítani az adott socketen a SO_NOSIGPIPE flag-et, majd write() vagy más
függvénnyel lehet a socketre irkálni. További probléma a netsocket
szerkezetéből adódik, ahová mindig eljutok, amikor aszinkron rendszerben
meghívódik egy callback és a callback-en belül van törölve az az objektum,
ami a callback-et meghívta. Ez ugye paradoxon, avagy magad alatt vágod a
fát, C-ben pedig csúnya segfault vagy még rosszabb. Jelen esetben itt a
send() függvény visszatérési értékénél kellene ellenőrizni a "Broken pipe"
esetet és ha bekövetkezik, akkor a netsocket_disconnect() függvénnyel
megszakítani a kapcsolatot és hátrahagyni a "Broken pipe" üzenetet. Ebben az
esetben a netsocket_disconnect meghívja a sock_close függvényt, ami törli a
netsocket objektumot, majd visszatér ide. Ezen a ponton már a netsocket
objektum, amivel eddig dolgoztunk, nem létezik és elképzelhető, hogy érkezik
még egy netsocket_write függvényhívás vagy eleve mivel callback-ben vagyunk,
más is történik a netsocket-tel, ami már végzetes egy nem létező objektumon.
Erre kell kitalálni valami megoldást! Két ötlet van: vagy hardcore módon
minden netsocket függvény a meghívásnál leellenőrzi, hogy a paraméterül
átadott netsocket objektum létezik, írható, stb., és ha nem, akkor visszatér
hibával. A másik módszer pedig, hogy aszinkron módon történik a netsocket
objektumok törlése. A netsocket_disconnect megrendeli a törlést, és majd a
callback lefutása után ténylegesen le is lesz törölve. Ezt az aszinkron
megoldást már csináltam Perl-ben és jól működött, de valahogyan most ezt a
kommentet fogalmazva, a hardcore verzió tűnik a legjobbnak, tehát: minden
azonnal történjen meg, semmi ne blokkolódjon, ha már nem létezik az adott
objektum, akkor hibával térjünk vissza és ne folytassuk a műveletet. Ezt
minden olyan helyen ellenőrizni kell, ahol az objektummal munka van. Ez a
koncepció talán a szálaknál is alkalmazható egy-egy mutex lock körzet alatt.
Végiggondolni a szitut úgy, hogy egy callback-ből hívott netsocket_write
hívja meg a netsocket_disconnect-et:) Praktikusan az, ami miatt ezt a
litániát megírtam. */
/* Ötlet. Az aszinkron megoldást végiggondolni jobban! A netsocket_destroy()
meghívására ne törlődjön azonnal a netsocket. Ha callback-ben van éppen,
akkor a callback után történjen a felszabadítás. Vagy dupla mutatót használni
és ellenőrizni a NULL értéket? Faszság:) */
// i = write(obj->sock, data, length); // nem szignál-biztos
i = send(obj->sock, data, length, MSG_NOSIGNAL);
return i;
}
/**
* @brief Create a new netsocket object
* @param callback callback function pointer
* @param userdata user defined pointer
* @param loop event loop
* @returns a new netsocket object
*
* Create and return a malloc'ed netsocket object. Save callback and userdata
* in the netsocket structure. Callback will be called when an event occur. Userdata
* is an user defined pointer. If the optional loop parameter is given, then tell
* the netsocket to use this event loop for async operations. This is useful in
* multithreading environment. If the loop is NULL, then the default event loop
* will be used.
*/
netsocket_t *netsocket_new (void *callback, void *userdata, struct ev_loop *loop) {
netsocket_t *obj = malloc(sizeof(*obj));
if (obj == NULL)
return NULL;
//~ printf("netsocket object size = %d\n", sizeof(*obj));
bzero(obj, sizeof(*obj)); // mindent nullázunk
// ha meg van adva a loop paraméter, akkor azt használjuk eseménykezelőnek
// ellenkező esetben az alapértelmezett eseménykezelőt
obj->loop = (loop != NULL) ? loop : ev_default_loop(0);
// default értékek
obj->connect_timeout = 5000; // 5000 millisec
obj->callback = callback;
obj->userdata = userdata;
obj->w_in.data = obj;
obj->w_out.data = obj;
obj->w_connect_timeout.data = obj;
ev_init(&obj->w_in, (void*)w_in_cb);
ev_init(&obj->w_out, (void*)w_out_cb);
ev_init(&obj->w_connect_timeout, (void*)w_connect_timeout_cb);
return obj;
}
/**
* @brief destroy the netsocket object
* @param obj the netsocket object
*
* Disconnect from peer(s) without call the callback, close sockets,
* free all resources and destroy the netsocket object.
*/
// TODO ha a szerver netsocket-et szabadítjuk fel, akkor az összes hozzá tartozó
// kliens is menjen a levesbe
void netsocket_destroy (netsocket_t *obj) {
// ha nincs objektum, akkor lófaszjóska
if (obj == NULL)
return;
if (obj->in_callback > 0) {
obj->destroy_request = 1;
return;
}
disconnect(obj, NULL, 1);
free(obj);
}
// formázott konzol üzenet kiírása időbélyeggel
int netsocket_printf (netsocket_t *obj, const char *fmt, ...) {
char tmp[8192];
va_list ap;
va_start(ap, fmt);
vsnprintf(tmp, sizeof(tmp) - 1, fmt, ap);
va_end(ap);
return netsocket_write(obj, tmp, strlen(tmp));
}
void netsocket_host (netsocket_t *obj, const char *host) {
if (host != NULL)
strncpy(obj->host, host, sizeof(obj->host) - 1);
}
void netsocket_lhost (netsocket_t *obj, const char *lhost) {
if (lhost != NULL)
strncpy(obj->lhost, lhost, sizeof(obj->lhost) - 1);
}
void netsocket_port (netsocket_t *obj, int port) {
obj->port = port;
}
void netsocket_lport (netsocket_t *obj, int lport) {
obj->lport = lport;
}
int netsocket_is_connected (netsocket_t *obj) {
return obj->connected;
}

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/* netsocket.h
* Copyright © 2014, Andras Jeszenszky, JSS & Hayer IT - http://www.jsshayer.hu
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#ifndef NETSOCKET_H_INCLUDED
#define NETSOCKET_H_INCLUDED
#include <netdb.h>
#include <ev.h>
#define NETSOCKET_IN 1
#define NETSOCKET_OUT 2
#define NETSOCKET_CLIENT 1
#define NETSOCKET_SERVER 2
#define NETSOCKET_TCP 1
#define NETSOCKET_UNIX 2
#define NETSOCKET_STATE_RESOLVING 1
#define NETSOCKET_STATE_CONNECTING 2
#define NETSOCKET_STATE_CONNECTED 3
#define NETSOCKET_EVENT_ERROR 1
#define NETSOCKET_EVENT_CONNECT 2
#define NETSOCKET_EVENT_READ 3
#define NETSOCKET_EVENT_DISCONNECT 4
typedef struct netsocket_t {
char host[128];
int port;
char lhost[128];
int lport;
int connect_timeout;
char ip[20];
int sock;
struct hostent *hostent;
struct sockaddr_in addr;
char inbuf[1024];
int inbuf_len;
ev_io w_out;
ev_io w_in;
ev_timer w_connect_timeout;
void (*callback)(void*, int);
void *userdata; // user data
int err;
int event;
char disconnect_reason[128];
int connected;
int mode;
struct netsocket_t *parent;
int direction;
int destroy_request;
int in_callback;
int disable_lookup_on_accept;
struct ev_loop *loop;
} netsocket_t;
netsocket_t *netsocket_new (void *callback, void *userdata, struct ev_loop *loop);
void netsocket_destroy (netsocket_t *obj);
int netsocket_connect (netsocket_t *obj);
int netsocket_listen (netsocket_t *obj);
void netsocket_disconnect (netsocket_t *obj, char *reason);
void netsocket_disconnect_withevent (netsocket_t *obj, char *reason);
int netsocket_write (netsocket_t *obj, const char *data, int length);
int netsocket_printf (netsocket_t *obj, const char *fmt, ...);
void netsocket_disable_lookup_on_accept (netsocket_t *obj);
void netsocket_host (netsocket_t *obj, const char *host);
void netsocket_lhost (netsocket_t *obj, const char *host);
void netsocket_port (netsocket_t *obj, int port);
void netsocket_lport (netsocket_t *obj, int port);
int netsocket_is_connected (netsocket_t *obj);
#endif // #ifndef NETSOCKET_H_INCLUDED

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