Package: asterisk / 1:16.2.1~dfsg-1+deb10u2
Metadata
| Package | Version | Patches format |
|---|---|---|
| asterisk | 1:16.2.1~dfsg-1+deb10u2 | 3.0 (quilt) |
Patch series
view the series file| Patch | File delta | Description |
|---|---|---|
| hack multiple app voicemail | (download) |
Makefile.moddir_rules |
2 1 + 1 - 0 ! |
build multiple versions of app_voicemail.so
This is a very ugly hack on upstream's Makefiles to allow building
multiple variants of app_voicemail. Three variants are created:
* app_voicemail.so: plain old filesystem storage that doesn't break
existing setups
* app_voicemail_imapstorage.so: IMAP storage
* app_voicemail_odbcstorage.so: ODBC storage
All these conflict with each other and Asterisk will refuse to load
them concurrently. They are thus included in three separate and
complicting packages.
.
Patch suggested to upstream but rejected for being "hackish". Though
upstream RPM packages include packages that are only somewhat cleaner.
|
| astgenkey security | (download) |
contrib/scripts/astgenkey |
4 4 + 0 - 0 ! |
astgenkey should generate a private key that is not world-readable Upstream has not accepted this patch and chose instead to document this as a known minor issue. |
| sound_files | (download) |
sounds/sounds.xml |
2 0 + 2 - 0 ! |
avoid downloading extra sound files Asterisk configures several sound files to be installed that are not included in the distribution tarball. Those files are downloaded by the 'install' target. . The exact files to be downloaded is configurable. Here we change the default to avoid downloading any. We believe those should be part of a separate source package (as they rarely change, and have their own versioning). |
| mpglib | (download) |
addons/mp3/MPGLIB_README |
39 39 + 0 - 0 ! |
mpglib code originally in asterisk-addons The package asterisk-addons originally included mpglib. After the merge with asterisk, that code is no longer included and needs to be fetched (contrib/scripts/get_mpg_source.sh). This patch includes that fetched source (rev. 202). . TODO: get rid of this code and use libmpg123 or whatever. |
| enable_addons | (download) |
addons/app_mysql.c |
1 0 + 1 - 0 ! |
enable modules formly from asterisk-addons The modules under addons/ are originally from the separate asterisk-addons package. As of asterisk 1.8 they are included in the main Asterisk distribution but not enabled by default. this patch enables them, as it seems valid in Debian. . format_mp3.c is not enabled, yet, though: the complete source is not included. See contrib/scripts/get_mp3_source.sh in the source tree. |
| ilbc_disable | (download) |
codecs/Makefile |
1 0 + 1 - 0 ! |
disable building codec_ilbc As we have to strip the ilbc code from asterisk, we need to disable building codec_ilbc and cleaning the ilbc/ directory. . Patch needs to be cleaned-up to be uploaded upstream. . FIXME: module now seems to potentially use libilbc. If it can be packaged into Debian, no reason to remove it. |
| astdatadir | (download) |
configure.ac |
2 1 + 1 - 0 ! |
place asterisk read-only data files under /usr/share On Debian read-only resources belong under /usr. The space taken from the writable /var should be minimized. . Upstream prefers defaults to have those files under /var/lib, though supports a separate datadir. |
| reenable | (download) |
channels/chan_mgcp.c |
1 0 + 1 - 0 ! |
reenable some drivers |
| no_native_arch.patch | (download) |
build_tools/cflags.xml |
2 1 + 1 - 0 ! |
disable building asterisk with -march=native Bug-Debian: https://bugs.debian.org/842917 |
| smsq_enable.patch | (download) |
utils/utils.xml |
2 1 + 1 - 0 ! |
enable the smsq application. |
| aelparse_enable.patch | (download) |
utils/utils.xml |
2 1 + 1 - 0 ! |
enable the aelparse application. |
| systemd.patch | (download) |
Makefile |
2 2 + 0 - 0 ! |
a systemd service Do away with safe_asterisk. But try very hard to let live_ast work with it. |
| test_framework.patch | (download) |
build_tools/cflags-devmode.xml |
3 0 + 3 - 0 ! |
enable the test framework |
| amr.patch | (download) |
build_tools/menuselect-deps.in |
3 3 + 0 - 0 ! |
add amr and amr-wb codec modules supporting transcoding To add a codec for SIP/SDP (m=, rtmap, and ftmp), you create a format module in Asterisk: `codec_amr.patch` (for m= and rtmap) and `res/res_format_attr_amr.c` (for fmtp). However, this requires both call legs to support AMR (pass-through only). If one leg does not support AMR, the call has no audio. Or, if you use the pre-recorded voice and music files of Asterisk, these files cannot be heard, because they are not in AMR but in slin. Therefore, this repository adds not just a format module for the audio-codecs AMR and AMR-WB but a transcoding module as well: `codecs/codec_amr.c`. . This is an implementation of IETF [RFC 4867](http://tools.ietf.org/html/rfc4867). Sometimes, AMR is called AMR Narrowband (AMR-NB). AMR Wideband (ITU-T Recommendation G.722.2) is sometimes abbreviated W-AMR ([GSA](http://www.gsacom.com/hdvoice/)). GSMA Mobile [HD Voice](https://www.youtube.com/playlist?&list=PLj1MyDu3jckpSciPQ1Max0W6HDSaY8-n4) is AMR-WB. Research papers comparing AMR and AMR-WB with other audio codecs: [InterSpeech 2010](http://research.nokia.com/files/public/%5B12%5D_Interspeech%202010_Voice%20Quality%20Evaluation%20of%20Recent%20Open%20Source%20Codecs.pdf), [ICASSP 2010](http://research.nokia.com/files/public/%5B11%5D_ICASSP2010_Voice%20Quality%20Evaluation%20of%20Various%20Codecs.pdf), [InterSpeech 2011](http://research.nokia.com/files/public/%5B16%5D_InterSpeech2011_Voice_Quality_Characterization_of_IETF_Opus_Codec.pdf). Further [examples](http://www.voiceage.com/Audio-Samples-Listening-Room.html) |
| ffmpeg detection.patch | (download) |
configure.ac |
15 13 + 2 - 0 ! |
modernize autotools ffmpeg linking FFmpeg is a _family_ of libraries sharing an optional base subdir. That is not properly reflected in the autoconf detection logic, and makes it impossible to handle alternate location - e.g. when using Libav. . This patch queries pkg-config, used with recent FFmpeg, for files "libavcodec" and "libswscale", the family members currently used. |
| ffmpeg includes.patch | (download) |
channels/console_video.h |
4 2 + 2 - 0 ! |
include subdirs (not main dir) for ffmpeg paths Fix include FFmpeg headers from below /usr/include/ffmpeg/<libname> (this change requires -I/usr/include/ffmpeg). |
| build reproducibly | (download) |
Makefile |
2 1 + 1 - 0 ! |
--- |
| autoreconf pjproject | (download) |
third-party/pjproject/Makefile |
5 5 + 0 - 0 ! |
update autoconf files for pjproject config.guess and config.sub for pjproject are six years old, this makes the build FTBFS on newer architectures like ppc64el. . Unfortunately the sources are only unpacked during the toplevel ./configure run, so we cannot solve this with dh_autoreconf |
| AST 2019 002.patch | (download) |
res/res_pjsip_messaging.c |
9 6 + 3 - 0 ! |
[patch] res_pjsip_messaging: check for body in in-dialog message We now check that a body exists and it has a length > 0 before attempting to process it. ASTERISK-28447 Reported-by: Gil Richard |
| AST 2019 003.patch | (download) |
channels/chan_sip.c |
8 7 + 1 - 0 ! |
[patch] chan_sip: handle invalid sdp answer to t.38 re-invite The chan_sip module performs a T.38 re-invite using a single media stream of udptl, and expects the SDP answer to be the same. If an SDP answer is received instead that contains an additional media stream with no joint codec a crash will occur as the code assumes that at least one joint codec will exist in this scenario. This change removes this assumption. ASTERISK-28465 |
| AST 2019 004.patch | (download) |
res/res_pjsip_t38.c |
72 45 + 27 - 0 ! |
[patch] ast-2019-004 - res_pjsip_t38.c: add null checks before using session media After receiving a 200 OK with a declined stream in response to a T.38 initiated re-invite Asterisk would crash when attempting to dereference a NULL session media object. This patch checks to make sure the session media object is not NULL before attempting to use it. ASTERISK-28495 patches: ast-2019-004.patch submitted by Alexei Gradinari (license 5691) |
| AST 2019 006.patch | (download) |
channels/chan_sip.c |
28 16 + 12 - 0 ! |
[patch] chan_sip.c: prevent address change on unauthenticated sip request. If the name of a peer is known and a SIP request is sent using that peer's name, the address of the peer will change even if the request fails the authentication challenge. This means that an endpoint can be altered and even rendered unusuable, even if it was in a working state previously. This can only occur when the nat option is set to the default, or auto_force_rport. This change checks the result of authentication first to ensure it is successful before setting the address and the nat option. ASTERISK-28589 #close |
| AST 2019 007.patch | (download) |
doc/UPGRADE-staging/AMI-Originate.txt |
5 5 + 0 - 0 ! |
[patch] manager.c: prevent the originate action from running the originate app If an AMI user without the "system" authorization calls the Originate AMI command with the Originate application, the second Originate could run the "System" command. Action: Originate Channel: Local/1111 Application: Originate Data: Local/2222,app,System,touch /tmp/owned If the "system" authorization isn't set, we now block the Originate app as well as the System, Exec, etc. apps. ASTERISK-28580 Reported by: Eliel Sardaons |
| fix error building json.patch | (download) |
res/res_pjsip_session.c |
4 3 + 1 - 0 ! |
[patch] res_pjsip_session.c: prevent use-after-free with test_framework enabled We need to copy the endpoint name before we call ao2_cleanup() on it, otherwise we might try to access memory that has been reclaimed. ASTERISK-28445 #close Reported by: Bernhard Schmidt |
| fix sigsegv pjsip history.patch | (download) |
res/res_pjsip_history.c |
4 2 + 2 - 0 ! |
[patch] res_pjsip_history.c: fix to stop sigsegv when ipv6 addresses are encountered. Changed source and destination address fields in struct pjsip_history_entry so that they are long enough to hold an IPv6 address. ASTERISK-28854 |
