kumquat-buildroot/package/asterisk/asterisk.hash

16 lines
1.0 KiB
Plaintext
Raw Normal View History

package/asterisk: new package Asterisk: the flagship of telephony on Linux. These are the lines of code whose continuous mission is to power small and large enterprises telephony systems, to boldly provide IP PBX where no one has done so before. But it is a hell to get compiled... :-( For starters, it needs a host tool, menuselect, to prepare its build configuration. Unfortunately, the way it handles menuselect does not apply very well for cross-compilation: the main ./configure calls out to menuselect's own ./configure, and of course that runs with the same environement, which is wrong for cross-compilation (because of variables like CC, CFLAGS and the likes). Furthermore, the paths to menuselect are imbricated about everywhere in the main Makefile, so making it find menuselect in PATH is a lost cause. Instead, we just patch-out the handling of menuselect, build it as the host variant and copy it in place. Now, asterisk wants to install a default set of sound files (for answering machine stuff, I guess). They come come pre-bundled in the official archive [0], but the buildsystem will want to download (at install time) the sha1 files for each sound archive, to validate that said archive is correct. However, the download is done via plain http, so it still risks an MITM attack. And for Buildroot, it is not always possible to download at install time, so we patch-out the sha1 check. [0] http://downloads.asterisk.org/pub/telephony/asterisk/releases/ The official archive contains the sound archives plus a full set of documentation. This makes it very big. Unfortunately, the hosting site is rather slow, topping at about ~204kbps. So we get the archive from the official mirror on Github. But that archive is missing the sound archives, so we download them separately. Some tests, like the crypt() one, are broken and could not have ever possibly worked at all. Worse, the FFmpeg test is looking for headers that FFmpeg removed more than 10 years ago and are virtually no longer available in any distro. So, FFmpeg support is definitely not tested by upstream and can't possibly work at all. Finally, trying to run test-code does not work in cross-compilation. As a final stroke of genius, asterisk checks for the re-entrant variant of res_ninit(), and concludes that all such functions are available, including res_nsearch(). Uclibc-ng has the former but not the latter, so the build fails. Since there is no cache variable for that check, we can't pre-feed that result to configure, and fixing it is a bigger endeavour. So we make asterisk depend on glibc for now, until someone is brave enough to fix it. Almost all features are disabled for now. Support for additional features will be added in subsequent patches now that we have a working base. Signed-off-by: "Yann E. MORIN" <yann.morin.1998@free.fr> Cc: Romain Naour <romain.naour@openwide.fr> Cc: Thomas Petazzoni <thomas.petazzoni@free-electrons.com> [Arnout: - make libilbc a mandatory dependency instead of using the bundled one; - add license, license files, and license file hashes; - minor spelling corrections; - remove redundant trailing backslash reported by check-package; - rewrap help text to 72 columns instead of 68] Signed-off-by: Arnout Vandecappelle (Essensium/Mind) <arnout@mind.be> fixup
2017-09-09 23:39:07 +02:00
# Locally computed
asterisk: security bump to version 14.6.2 Fixes the following security issues: 14.6.1: * AST-2017-005 (applied to all released versions): The "strictrtp" option in rtp.conf enables a feature of the RTP stack that learns the source address of media for a session and drops any packets that do not originate from the expected address. This option is enabled by default in Asterisk 11 and above. The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip respectively enable symmetric RTP support in the RTP stack. This uses the source address of incoming media as the target address of any sent media. This option is not enabled by default but is commonly enabled to handle devices behind NAT. A change was made to the strict RTP support in the RTP stack to better tolerate late media when a reinvite occurs. When combined with the symmetric RTP support this introduced an avenue where media could be hijacked. Instead of only learning a new address when expected the new code allowed a new source address to be learned at all times. If a flood of RTP traffic was received the strict RTPsupport would allow the new address to provide media and with symmetric RTP enabled outgoing traffic would be sent to this new address, allowing the media to be hijacked. Provided the attacker continued to send traffic they would continue to receive traffic as well. * AST-2017-006 (applied to all released versions): The app_minivm module has an “externnotify” program configuration option that is executed by the MinivmNotify dialplan application. The application uses the caller-id name and number as part of a built string passed to the OS shell for interpretation and execution. Since the caller-id name and number can come from an untrusted source, a crafted caller-id name or number allows an arbitrary shell command injection. * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in a From, To or Contact header could cause Asterisk to crash For more details, see the announcement: https://www.asterisk.org/downloads/asterisk-news/asterisk-11252-13171-1461-116-cert17-1313-cert5-now-available-security 14.6.2: * AST-2017-008: Insufficient RTCP packet validation could allow reading stale buffer contents and when combined with the “nat” and “symmetric_rtp” options allow redirecting where Asterisk sends the next RTCP report. The RTP stream qualification to learn the source address of media always accepted the first RTP packet as the new source and allowed what AST-2017-005 was mitigating. The intent was to qualify a series of packets before accepting the new source address. For more details, see the announcement: https://www.asterisk.org/downloads/asterisk-news/asterisk-11253-13172-1462-116-cert18-1313-cert6-now-available-security Drop 0004-configure-in-cross-complation-assimne-eventfd-are-av.patch as this is now handled differently upstream (by disabling eventfd for cross compilation, see commit 2e927990b3d2 (eventfd: Disable during cross compilation)). If eventfd support is needed then this should be submitted upstream. Signed-off-by: Peter Korsgaard <peter@korsgaard.com> Reviewed-by: "Yann E. MORIN" <yann.morin.1998@free.fr> Signed-off-by: Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
2018-01-07 22:46:29 +01:00
sha256 f85f6df802de485d9b8cb1bfa5493e22f6401dce8246646af9506489a264d7b1 asterisk-14.6.2.tar.gz
package/asterisk: new package Asterisk: the flagship of telephony on Linux. These are the lines of code whose continuous mission is to power small and large enterprises telephony systems, to boldly provide IP PBX where no one has done so before. But it is a hell to get compiled... :-( For starters, it needs a host tool, menuselect, to prepare its build configuration. Unfortunately, the way it handles menuselect does not apply very well for cross-compilation: the main ./configure calls out to menuselect's own ./configure, and of course that runs with the same environement, which is wrong for cross-compilation (because of variables like CC, CFLAGS and the likes). Furthermore, the paths to menuselect are imbricated about everywhere in the main Makefile, so making it find menuselect in PATH is a lost cause. Instead, we just patch-out the handling of menuselect, build it as the host variant and copy it in place. Now, asterisk wants to install a default set of sound files (for answering machine stuff, I guess). They come come pre-bundled in the official archive [0], but the buildsystem will want to download (at install time) the sha1 files for each sound archive, to validate that said archive is correct. However, the download is done via plain http, so it still risks an MITM attack. And for Buildroot, it is not always possible to download at install time, so we patch-out the sha1 check. [0] http://downloads.asterisk.org/pub/telephony/asterisk/releases/ The official archive contains the sound archives plus a full set of documentation. This makes it very big. Unfortunately, the hosting site is rather slow, topping at about ~204kbps. So we get the archive from the official mirror on Github. But that archive is missing the sound archives, so we download them separately. Some tests, like the crypt() one, are broken and could not have ever possibly worked at all. Worse, the FFmpeg test is looking for headers that FFmpeg removed more than 10 years ago and are virtually no longer available in any distro. So, FFmpeg support is definitely not tested by upstream and can't possibly work at all. Finally, trying to run test-code does not work in cross-compilation. As a final stroke of genius, asterisk checks for the re-entrant variant of res_ninit(), and concludes that all such functions are available, including res_nsearch(). Uclibc-ng has the former but not the latter, so the build fails. Since there is no cache variable for that check, we can't pre-feed that result to configure, and fixing it is a bigger endeavour. So we make asterisk depend on glibc for now, until someone is brave enough to fix it. Almost all features are disabled for now. Support for additional features will be added in subsequent patches now that we have a working base. Signed-off-by: "Yann E. MORIN" <yann.morin.1998@free.fr> Cc: Romain Naour <romain.naour@openwide.fr> Cc: Thomas Petazzoni <thomas.petazzoni@free-electrons.com> [Arnout: - make libilbc a mandatory dependency instead of using the bundled one; - add license, license files, and license file hashes; - minor spelling corrections; - remove redundant trailing backslash reported by check-package; - rewrap help text to 72 columns instead of 68] Signed-off-by: Arnout Vandecappelle (Essensium/Mind) <arnout@mind.be> fixup
2017-09-09 23:39:07 +02:00
# sha1 from: http://downloads.asterisk.org/pub/telephony/sounds/releases
# sha256 locally computed
sha1 65ee068462c6645ed14a28d6b34eb0e9aa7a6c8d asterisk-core-sounds-en-gsm-1.5.tar.gz
sha256 8d1118c6e0a0c614fafe297e3789f924ef5b04285cf6a8cffb8501dfcf5bbf07 asterisk-core-sounds-en-gsm-1.5.tar.gz
sha1 f40fd6ea03dfe8d72ada2540b2288bfdc006381d asterisk-moh-opsound-wav-2.03.tar.gz
sha256 449fb810d16502c3052fedf02f7e77b36206ac5a145f3dacf4177843a2fcb538 asterisk-moh-opsound-wav-2.03.tar.gz
# License files, locally computed
sha256 82af40ed7f49c08685360811993d9396320842f021df828801d733e8fdc0312f COPYING
sha256 ac5571f00e558e3b7c9b3f13f421b874cc12cf4250c4f70094c71544cf486312 main/sha1.c
sha256 0fcdb946955d20c2819a51f3fe613d8f22da2ea793bd50acb30ce156499acc88 codecs/speex/speex_resampler.h
sha256 e6e7b7204d34a3dcdf17389a9c8cf64721ec0d15a797fd51c8c1ed8517cc3038 utils/db1-ast/include/db.h