asterisk

Commit Graph

Author	SHA1	Message	Date
Ben Ford	a84d34035a	STIR/SHAKEN: Switch to base64 URL encoding. STIR/SHAKEN encodes using base64 URL format. Currently, we just use base64. New functions have been added that convert to and from base64 encoding. The origid field should also be an UUID. This means there's no reason to have it as an option in stir_shaken.conf, as we can simply generate one when creating the Identity header. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c	4 years ago
Ben Ford	5e6508b56f	STIR/SHAKEN: Fix certificate type and storage. During OpenSIPit, we found out that the public certificates must be of type X.509. When reading in public keys, we use the corresponding X.509 functions now. We also discovered that we needed a better naming scheme for the certificates since certificates with the same name would cause issues (overwriting certs, etc.). Now when we download a public certificate, we get the serial number from it and use that as the name of the cached certificate. The configuration option public_key_url in stir_shaken.conf has also been renamed to public_cert_url, which better describes what the option is for. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d	4 years ago
Asterisk Development Team	1949d828b7	Update CHANGES and UPGRADE.txt for 18.4.0	5 years ago
Jaco Kroon	f47c5cbdf9	menuselect: exit non-zero in case of failure on --enable\|disable options. ASTERISK-29348 Change-Id: I77e3466435f5a51a57538b29addb68d811af238d Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Asterisk Development Team	0afd37e3b5	Update CHANGES and UPGRADE.txt for 18.3.0	5 years ago
Alexander Traud	703158b903	rtp: Enable srtp replay protection Add option "srtpreplayprotection" rtp.conf to enable srtp replay protection. ASTERISK-29260 Reported by: Alexander Traud Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458	5 years ago
Asterisk Development Team	1f5e6805bf	Update CHANGES and UPGRADE.txt for 18.0.0	5 years ago
Walter Doekes	312c23b0e1	app_queue: (Breaking change) shared_lastcall and autofill default to no If your queues.conf had _no_ [general] section, they would default to 'yes'. Now, they always default to 'no'. (Actually, commit `ed615afb7e` already partially fixed it for shared_lastcall.) ASTERISK-28951 Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6	5 years ago
George Joseph	8d1064eaaf	Streams: Add features for Advanced Codec Negotiation The Streams API becomes the home for the core ACN capabilities. These include... * Parsing and formatting of codec negotation preferences. * Resolving pending streams and topologies with those configured using configured preferences. * Utility functions for creating string representations of streams, topologies, and negotiation preferences. For codec negotiation preferences: * Added ast_stream_codec_prefs_parse() which takes a string representation of codec negotiation preferences, which may come from a pjsip endpoint for example, and populates a ast_stream_codec_negotiation_prefs structure. * Added ast_stream_codec_prefs_to_str() which does the reverse. * Added many functions to parse individual parameter name and value strings to their respectrive enum values, and the reverse. For streams: * Added ast_stream_create_resolved() which takes a "live" stream and resolves it with a configured stream and the negotiation preferences to create a new stream. * Added ast_stream_to_str() which create a string representation of a stream suitable for debug or display purposes. For topology: * Added ast_stream_topology_create_resolved() which takes a "live" topology and resolves it, stream by stream, with a configured topology stream and the negotiation preferences to create a new topology. * Added ast_stream_topology_to_str() which create a string representation of a topology suitable for debug or display purposes. * Renamed ast_format_caps_from_topology() to ast_stream_topology_get_formats() to be more consistent with the existing ast_stream_get_formats(). Additional changes: * A new function ast_format_cap_append_names() appends the results to the ast_str buffer instead of replacing buffer contents. Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56	5 years ago
Ben Ford	9acf840f7c	res_stir_shaken: Implemented signature verification. There are a lot of moving parts in this patch, but the focus of it is on the verification of the signature using a public key located at the public key URL provided in the JSON payload. First, we check the database to see if we have already downloaded the key. If so, check to see if it has expired. If it has, redownload from the URL. If we don't have an entry in the database, just go ahead and download the public key. The expiration is tested each time we download the file. After that, read the public key from the file and use it to verify the signature. All sanity checking is done when the payload is first received, so the verification is complete once this point is reached. The XML has also been added since a new config option was added to general (curl_timeout). The maximum amount of time to wait for a download can be configured through this option, with a low value by default. Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c	6 years ago
Joshua C. Colp	1c5e68580a	stream: Enforce formats immutability and ensure formats exist. Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	6 years ago
Kevin Harwell	a715cf5aaa	message & stasis/messaging: make text message variables work in ARI When a text message was received any associated variable was not written to the ARI TextMessageReceived event. This occurred because Asterisk only wrote out "send" variables. However, even those "send" variables would fail ARI validation due to a TextMessageVariable formatting bug. Since it seems the TextMessageReceived event has never been able to include actual variables it was decided to remove the TextMessageVariable object type from ARI, and simply return a JSON object of key/value pairs for variables. This aligns more with how the ARI sendMessage handles variables, and other places in ARI. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	6 years ago
Sean Bright	8dcdce42a9	app_mixmonitor: Turn on synchronization by default The optional synchronization behavior created in `64906c4c9b` is now the default for MixMonitor. * Add a new flag 'n' that allows for this behavior to be turned off * Add a notice when the 'S' option is used indicating that it is no longer necessary Change-Id: I158987c475cda4e1ff1256dd0daccdd99df568b4	6 years ago
George Joseph	a72caa041f	doc: Fix CHANGES entries to have .txt suffix and update READMEs Although the wiki page for the new CHANGES and UPGRADE scheme states that the files must have the ".txt" suffix, the READMEs didn't. Change-Id: I490306aa2cc24d6f014738e9ebbc78592efe0f05 (cherry picked from commit `7416703f04`)	6 years ago
Sean Bright	f09cf4da44	app_voicemail: Remove MessageExists and MESSAGE_EXISTS() * The MailboxExists dialplan application was deprecated on 2006-09-26 in Asterisk 1.6.0 (commit `ec83b11183`) * The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in Asterisk 11.0.0 (commit `fd64bb66f9`) Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f	6 years ago
Sean Bright	9522390a69	app_queue: Deprecate the QueueMemberPause.Reason field The QueueMemberPause AMI event includes two fields that return the reason a member was paused. * In release branches, deprecate Reason in favor of PausedReason. * In master, remove the Reason field entirely. ASTERISK-28349 #close Reported by: Niksa Baldun Change-Id: I01da58f2b0ab927baeee754870f62b51b7b3d296	6 years ago
Richard Mudgett	19069f7db7	app_bridgeaddchan.c: Make BridgeAdd be more like Bridge * Made BridgeAdd not hangup the call if there is a problem. * Reduced message level from warning to verbose for normal exception cases. * Added a loop safety check to BridgeAdd. * Made BridgeAdd set BRIDGERESULT with the status when dialplan is resumed. Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426	6 years ago
George Joseph	7e3a6e158f	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 Sardañons Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa	6 years ago
Asterisk Development Team	5e6e1175d5	Update CHANGES and UPGRADE.txt for 17.0.0	6 years ago
George Joseph	799c4cf494	Merge "chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS"	6 years ago
George Joseph	c781806e26	Build: Separate header install/uninstall Asterisk headers are no longer installed and uninstalled automatically when performing a "make install" or a "make uninstall". To install/uninstall the headers, use "make install-headers" and "make uninstall-headers". The headers also continue to be uninstalled when performing a "make uninstall-all". Also corrects an issue where /usr/include/asterisk.h was never being removed at all. Change-Id: Ia7399f3a0203a4825fc4a9f43b9034dae9a2b643	6 years ago
Chris-Savinovich	6b1f6ea2c4	app_voicemail.c: Build all three variants for app_voicemail at the same time Changes made to apps/Makefile to optionally build all three app_voicemail variations at the same time: 1) file (default), 2) odbc, and 3) imap. This functionality was requested by users. modules.conf.sample warns the user to make sure only one voicemail is loaded at a time. Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7	6 years ago
Dan Cropp	e52fbae00f	chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS Previously, when a Transfer (REFER) was performed, chan_pjsip would set the TRANSFERSTATUS to SUCCESS when the REFER was queued up. This did not reflect a successful/unsuccessful transfer the way chan_sip did. Added a callback module to process the refer subscription information. Now depends on res_pjsip_pubsub so call transfer progress can be monitored and reported ASTERISK-26968 #close Reported-by: Dan Cropp Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc	6 years ago
Ben Ford	a4ab7f5f80	build: Revise CHANGES and UPGRADE.txt handling. This changes the way that we handle adding changes to CHANGES and UPGRADE.txt. The reason for this is because whenever someone needed to make a change to one of these files and someone else had already done so, you would run into merge conflicts. With this new setup, there will never be merge conflicts since all changes will be documented in the doc/<file>-staging directory. The release script is now responsible for merging all of these changes into the appropriate files. There is a special format that these files have to follow in order to be parsed. The files do not need to have a meaningful name, but it is strongly recommended. For example, if you made a change to pjsip, you may have something like this "res_pjsip_relative_title", where "relative_title" is something more descriptive than that. Inside each file, you will need a subject line for your change, followed by a description. There can be multiple subject lines. The file may look something like this: Subject: res_pjsip Subject: Core A description that explains the changes made and why. The release script will handle the bulleting and section separators! You can still separate with new lines within your description. The headers ("Subject" and "Master-Only") are case sensative, but the value for "Master-Only" ("true" or "True") is not. For more information, check out the wiki page: https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt ASTERISK-28111 #close Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47	7 years ago
Ben Ford	d5d8448ce5	build: Add staging directories for future changes. This is the first step in changing the release process so that changes made to the CHANGES and UPGRADE.txt files do not result in merge conflicts every time multiple people modify these files. The changes made will go in these new directories: doc/CHANGES-staging and doc/UPGRADE-staging. The README.md files explain how things will work, but here's a little overview. When you make a change that would go in either CHANGES or UPGRADE.txt, this should instead be documented in a new file in the doc/CHANGES-staging or doc/UPGRADE-staging directory, respectively. The format will look like this: Subject: res_pjsip A description that explains the changes made and why. The release script will handle the bulleting and section separators! The 'Subject:' header is case-sensitive. You can still separate with new lines within your description. Subject: res_ari Master-Only: true You can have more than one subject, and they don't have to be the same! Also, the 'Master-Only' header should always be true and is also case-sensitive (but the value is not - you can have 'true' or 'True'). This header will only ever be present in the master branch. For more information, check out the wiki page: https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt This is an initial change for ASTERISK_28111. Functionally, this will make no difference, but it will prep the directories for when the changes from CHANGES and UPGRADE.txt are extracted. Change-Id: I8d852f284f66ac456b26dcb899ee46babf7d15b6	7 years ago

25 Commits (bfc25e5de2d8f75df37b65a966cf7fd00c5be43d)