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10.12.2-digiumphones
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10.12.3-digiumphones
10.12.4
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certified/1.8.11-cert1
certified/1.8.11-cert10
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certified/1.8.11-cert3-rc1
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certified/1.8.11-cert6
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certified/1.8.15-cert1
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certified/1.8.28-cert1
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certified/1.8.28-cert4
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certified/1.8.6-cert1
certified/11.2-cert1
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certified/11.2-cert2
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certified/11.6-cert1
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certified/11.6-cert10
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certified/11.6-cert6
certified/11.6-cert7
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certified/13.1-cert1
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certified/13.1-cert1-rc2
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certified/13.1-cert2
certified/13.1-cert3
${ noResults }
87 Commits (d60bcc56a7dbba38f5a3a72a8b2455f8de7cfdcd)
Author | SHA1 | Message | Date |
---|---|---|---|
|
82bef2ed90 |
docs: Indent <since> tags.
Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0). |
3 months ago |
|
198b01c536 |
docs: Add version information to application and function XML elements
* Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml |
3 months ago |
|
316693ded9 |
docs: Add version information to configObject and configOption XML elements
Most of the configObjects and configOptions that are implemented with ACO or Sorcery now have `<since>/<version>` elements added. There are probably some that the script I used didn't catch. The version tags were determined by the following... * Do a git blame on the API call that created the object or option. * From the commit hash, grab the summary line. * Do a `git log --grep <summary>` to find the cherry-pick commits in all branches that match. * Do a `git patch-id` to ensure the commits are all related and didn't get a false match on the summary. * Do a `git tag --contains <commit>` to find the tags that contain each commit. * Weed out all tags not <major>.<minor>.0. * Sort and discard any <major>.0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the API was last touched. configObjects and configOptions elements implemented with the base ast_config APIs were just not possible to find due to the non-deterministic way they are accessed. Also note that if the API call was on modified after it was added, the version will be the one it was last modified in. Final note: The configObject and configOption elements were introduced in 12.0.0 so options created before then may not have any XML documentation. |
3 months ago |
|
468601fd13 |
manager: Add `<since>` tags for all AMI actions.
|
3 months ago |
|
a85f2bf34d |
res: Fix for Doxygen.
These are the remaining issues found in /res. ASTERISK-29761 Change-Id: I572e6019c422780dde5ce8448b6c85c77af6046d |
3 years ago |
|
cc025026b7 |
progdocs: Fix for Doxygen, the hidden parts.
ASTERISK-29779 Change-Id: If338163488498f65fa7248b60e80299c0a928e4b |
3 years ago |
|
24a04054ad |
documentation: Standardize examples
Most examples in the XML documentation use the example tag to demonstrate examples, which gets parsed specially in the Wiki to make it easier to follow for users. This fixes a few modules to use the example tag instead of vanilla para tags to bring them in line with the standard syntax. ASTERISK-29777 #close Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b |
3 years ago |
|
845ece8bc4 |
res_xmpp: Fix for Doxygen.
ASTERISK-29749 Change-Id: I7885793b63bdeaa883e76edb899bbba9660eb1c5 |
3 years ago |
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9ae9893c63 |
res: Spelling fixes
Correct typos of the following word families: identifying structures actcount initializer attributes statement enough locking declaration userevent provides unregister session execute searches verification suppressed prepared passwords recipients event because brief unidentified redundancy character the module reload operation backslashes accurate incorrect collision initializing instance interpreted buddies omitted manually requires queries generator scheduler configuration has owner resource performed masquerade apparently routable ASTERISK-29714 Change-Id: I88485116d2c59b776aa2e1f8b4ce8239a21decda |
3 years ago |
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09303e8e22 |
Updates for the MessageSend Dialplan App
Enhancements: * The MessageSend dialplan application now takes an optional third argument that can set the message's "To" field on outgoing messages. It's an alternative to using the MESSAGE(to) dialplan function. NOTE: No channel driver currently implements this field. A follow-on commit for res_pjsip_messaging will implement it for the chan_pjsip channel driver. * To prevent confusion with the first argument, currently named "to", it's been renamed to "destination". Its function, creating the request URI, hasn't changed. * The documentation for MessageSend was updated to be more clear about the parameters and how they interact the MESSAGE() dialplan function. * With the rename of MessageSend's first parameter, and the fact that message.c references <info> elements in chan_sip.c, res_pjsip_messaging.c and res_xmpp, they each needed documentation updates to use MessageDestinationInfo instead of MessageToInfo. * appdocsxml.dtd was updated to include a missing element declaration for "dataType". This was showing up as an error in Eclipse's dtd editor. * Despite the changes in this commit, there should be no impact to current users of MessageSend. Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a |
4 years ago |
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ff0d0ac23a |
mwi core: Move core MWI functionality into its own files
There is enough MWI functionality to warrant it having its own 'c' and header files. This patch moves all current core MWI data structures, and functions into the following files: main/mwi.h main/mwi.c Note, code was simply moved, and not modified. However, this patch is also in preparation for core MWI changes, and additions to come. Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0 |
6 years ago |
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021ce938ca
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astobj2: Remove legacy ao2_container_alloc routine.
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. Remove ao2_container_alloc macro. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088 |
7 years ago |
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3077ad0c24 |
stasis: Add internal filtering of messages.
This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190 |
7 years ago |
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9e8d671658 |
res_xmpp: Remove deprecated JabberStatus application.
Change-Id: I1a00ca22d59d6b6d2166aa56f0e9338a33e5ac60 |
7 years ago |
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bf2d35931d |
aco: Minimize use of regex.
Remove nearly all use of regex from ACO users. Still remaining: * app_confbridge has a legitamate use of option name regex. * ast_sorcery_object_fields_register is implemented with regex, all callers use simple prefix based regex. I haven't decided the best way to fix this in both 13/15 and master. Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b |
7 years ago |
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ee08f10d06 |
Fix ast_(v)asprintf() malloc failure usage conditions.
When (v)asprintf() fails, the state of the allocated buffer is undefined. The library had better not leave an allocated buffer as a result or no one will know to free it. The most likely way it can return failure is for an allocation failure. If the printf conversion fails then you actually have a threading problem which is much worse because another thread modified the parameter values. * Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL on failure. That is much more useful than either an uninitialized pointer or a pointer that has already been freed. Many uses won't have to check for failure to ensure that the buffer won't be double freed or prevent an attempt to free an uninitialized pointer. * stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by ast_asprintf(). * ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to the wrong thing which is now not needed even if assigning to the right thing. Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23 |
8 years ago |
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252353e0a9 |
res_xmpp: Ensure the connection filter is available.
Users of the API that res_xmpp provides expect that a filter be available on the client at all times. When OAuth authentication support was added this requirement was not maintained. This change merely moves the OAuth authentication to after the filter is created, ensuring users of res_xmpp can add things to the filter as needed. ASTERISK-27346 Change-Id: I4ac474afe220e833288ff574e32e2b9a23394886 |
8 years ago |
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83b81d1f8d |
res_xmpp: fix inverted return code check in OAuth
fetch_access_token calls func_curl via ast_func_read. The latter returns 0 upon success and -1 if the function is not available. This commit inverts the return code check so that an error is printed if the module is not loaded and not if it is loaded. ASTERISK-27207 #close Change-Id: I9ef903f80702d1218e8701f65a4e5e918e6548fb |
8 years ago |
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15fbcc74d8 |
res_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif
Add ability to use tokens instead of passwords according to Google OAuth 2.0 protocol. ASTERISK-27169 Reported by: Andrey Egorov Tested by: Andrey Egorov Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db |
8 years ago |
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f43cfb81d9 |
Merge "res_xmpp: Fix ref counting issue"
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8 years ago |
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d5a8799c4b |
res_xmpp: Use incremental backoff when a read error occurs
If a read error occurs, we immediately attempt a reconnect without any delay. Instead, let's sleep and backoff up to 60 seconds before we try again. ASTERISK-24712 #close Reported by: Matthias Urlichs Change-Id: I6fe10ef4734837727437beab715e336777f13f48 |
8 years ago |
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f27c1c7dc2 |
Merge "res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus"
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8 years ago |
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8982d0e007 |
Merge "res_xmpp: Include client name in connection related error messages"
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8 years ago |
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6666deb907 |
Merge "res_xmpp: Don't crash when trying to send a message without a connection"
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8 years ago |
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1a626ffb89 |
Merge "res_xmpp: Correctly check return value of SSL_connect"
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8 years ago |
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98a88e9ffa |
res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus
The documentation for JABBER_STATUS (and the deprecated JabberStatus app) indicate that a return value of 7 indicates that the specified buddy was not in the roster. It also indicates that you can specify a "bare" JID (one without a resource). Unfortunately the actual behavior does not match the documented behavior. Assuming that our roster includes the buddy online and available "valid@example.org/Valid" and does *not* include the buddy "invalid@example.org", the JABBER_STATUS() function returns the following before this patch: +------------------------------+------------+--------------------------+ | Buddy | Status | Result | +------------------------------+------------+--------------------------+ | valid@example.org | Online | 7 (Not in roster) | | valid@example.org/Valid | Online | 1 (Online) | | valid@example.org/Invalid | N/A | 7 (Not in roster) | | invalid@example.org | N/A | Error logged, no return | | invalid@example.org/Valid | N/A | Error logged, no return | +------------------------------+------------+--------------------------+ And after this patch: +------------------------------+------------+--------------------------+ | Buddy | Status | Result | +------------------------------+------------+--------------------------+ | valid@example.org | Online | 1 (Online) | | valid@example.org/Valid | Online | 1 (Online) | | valid@example.org/Invalid | N/A | 6 (Offline) | | invalid@example.org | N/A | 7 (Not in roster) | | invalid@example.org/Valid | N/A | 7 (Not in roster) | +------------------------------+------------+--------------------------+ This brings the behavior in line with the documentation. ASTERISK-23510 #close Reported by: Anthony Critelli Change-Id: I9c3241035363ef4a6bdc21fabfd8ffcd9ec657bf |
8 years ago |
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be94105d6d |
res_xmpp: Try to provide useful errors messages from OpenSSL
If any errors occur during the TLS connection setup, we currently dump a fairly generic error message. So instead we try to pull in something useful from OpenSSL to report instead. ASTERISK-24712 Reported by: Matthias Urlichs Change-Id: I288500991a9681f447d92913b11fedaf426087f4 |
8 years ago |
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ee81ee1f14 |
res_xmpp: Fix ref counting issue
The only remaining reference to the endpoint is in the endpoints container, and because it is unlinked in ast_endpoint_shutdown, we don't have to explicitly cleanup the endpoint ourselves. Change-Id: I912a2692e52d3e2ed445b32d8ae3f9004bc2f2e8 |
8 years ago |
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9493981419 |
res_xmpp: Correctly check return value of SSL_connect
SSL_connect returns non-zero for both success and some error conditions so simply negating is inadequate. Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1 |
8 years ago |
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7657c279b5 |
res_xmpp: Don't crash when trying to send a message without a connection
If we never establish a connection to our Jabber server, iksemel never sets up its internal transport pointer, so attempting to send a message dereferences a NULL pointer and causes a crash. ASTERISK-21855 #close Reported by: Jeremy Kister Change-Id: I204a568894e4a53ab929783ecc594a000f04d79c |
8 years ago |
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0136ec12a3 |
res_xmpp: Include client name in connection related error messages
ASTERISK-25622 #close Reported by: Sean Darcy Change-Id: I8472cb7bfb58d411a3cfbd482da98cae2d94d1e9 |
8 years ago |
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a6e5bae3ef |
Remove ASTERISK_REGISTER_FILE.
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966 |
9 years ago |
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824a4e84d1 |
Refactor usage pattern of xmldoc info tag.
This updates func_channel.c and main/message.c to use a generic xpointer include instead of including info from each channel driver. Now the name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in documentation for func_channel. Setting the name attribute of info to MessageToInfo or MessageFromInfo causes it to be included in the MessageSend application and AMI action. Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea |
9 years ago |
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e85adbd947 |
core: Entity ID is not set or invalid
The Exchanging Device and Mailbox States could not working if the Entity ID (EID) is not set manually and can't be obtained from ethernet interface. This patch replaces debug message to warning and addes missing description about option 'entityid' to asterisk.conf.sample. With this patch the asterisk also: (1) decline loading the modules which won't work without EID: res_corosync and res_pjsip_publish_asterisk. (2) warn if EID is empty on loading next modules: pbx_dundi, res_xmpp Starting with v197 systemd/udev will automatically assign "predictable" names for all local Ethernet interfaces. This patch also addes some new ethernet prefixes "eno" and "ens". ASTERISK-26164 #close Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6 |
9 years ago |
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2a6f18cd55 |
res_xmpp: Does not connect in component mode
The module res_xmpp does not accept usernames in the form used in component mode (XEP-0114). In component mode there is no @something in the name. In component mode the connection is now not dropped anymore. If the xmpp server sends out a "stream" tag before handshake is finished, the connection gets dropped in res_xmpp. Now this tag will be ignored and the connection will be established. After connecting there will be an exchange of presence states. This does not work as expected in component mode. The responsible function "xmpp_pak_presence" is left before the states get sent out. Sending presence states in component mode is now moved to the top of the function. ASTERISK-25735 #close Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c |
9 years ago |
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50044fdc15 |
Stasis: Fix unsafe use of stasis_unsubscribe in modules.
Many uses of stasis_unsubscribe in modules can be reached through unload. These have been switched to stasis_unsubscribe_and_join. Some subscription callbacks do nothing, for these I've created a noop callback function in stasis.c. This is used by some modules that monitor MWI topics in order to enable cache, since the callback does not become invalid after dlclose it is safe to use stasis_unsubscribe on these, even during module unload. ASTERISK-25121 #close Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c |
10 years ago |
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eec010829a |
AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro.
Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723 |
10 years ago |
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4a58261694 |
git migration: Refactor the ASTERISK_FILE_VERSION macro
Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e |
10 years ago |
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47156aab92 |
res_xmpp: Buddies are always auto-registered when processing the roster
Due to a quirk in the configuration handling of res_xmpp, the 'autoregister' setting was never actually processed. This was due to not properly copying over the global settings to the client settings when applying the configuration to the run-time object. Review: https://reviewboard.asterisk.org/r/4496/ ASTERISK-14233 ASTERISK-24780 #close Reported by: Simon Arlott patches: asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756) ........ Merged revisions 433395 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433396 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433397 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
10 years ago |
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1106e8fd0f |
main/stasis: Allow subscriptions to use a threadpool for message delivery
Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428687 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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dad0334cf1 |
AST-2014-011: Fix POODLE security issues
There are two aspects to the vulnerability: (1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module to use TLSv1+. At this time, it does not refactor res_jabber/res_xmpp to use the TCP/TLS core, which should be done as an improvement at a latter date. (2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left unspecified, will default to the OpenSSL SSLv23_method. This method allows for all ecnryption methods, including SSLv2/SSLv3. A MITM can exploit this by forcing a fallback to SSLv3, which leaves the server vulnerable to POODLE. This patch adds WARNINGS if a user uses SSLv2/SSLv3 in their configuration, and explicitly disables SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly chosen. For TLS servers, Asterisk will no longer support SSLv2 or SSLv3. Much thanks to abelbeck for reporting the vulnerability and providing a patch for the res_jabber/res_xmpp modules. Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425 #close Reported by: abelbeck Tested by: abelbeck, opsmonitor, gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded by mjordan (License 6283) ........ Merged revisions 425987 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425991 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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47bf7efc4d |
Multiple revisions 420089-420090,420097
........ r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines ARI: Add channel technology agnostic out of call text messaging This patch adds the ability to send and receive text messages from various technology stacks in Asterisk through ARI. This includes chan_sip (sip), res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the endpoints resource, and can be sent directly through that resource, or to a particular endpoint. For example, the following would send the message "Hello there" to PJSIP endpoint alice with a display URI of sip:asterisk@mycooldomain.org: ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There This is equivalent to the following as well: ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There Both forms are available for message technologies that allow for arbitrary destinations, such as chan_sip. Inbound messages can now be received over ARI as well. An ARI application that subscribes to endpoints will receive messages from those endpoints: { "type": "TextMessageReceived", "timestamp": "2014-07-12T22:53:13.494-0500", "endpoint": { "technology": "PJSIP", "resource": "alice", "state": "online", "channel_ids": [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>", "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.", "variables": [] }, "application": "testsuite" } The above was made possible due to some rather major changes in the message core. This includes (but is not limited to): - Users of the message API can now register message handlers. A handler has two callbacks: one to determine if the handler has a destination for the message, and another to handle it. - All dialplan functionality of handling a message was moved into a message handler provided by the message API. - Messages can now have the technology/endpoint associated with them. Various other properties are also now more easily accessible. - A number of ao2 containers that weren't really needed were replaced with vectors. Iteration over ao2_containers is expensive and pointless when the lifetime of things is well defined and the number of things is very small. res_stasis now has a new file that makes up its structure, messaging. The messaging functionality implements a message handler, and passes received messages that match an interested endpoint over to the app for processing. Note that inadvertently while testing this, I reproduced ASTERISK-23969. res_pjsip_messaging was incorrectly parsing out the 'to' field, such that arbitrary SIP URIs mangled the endpoint lookup. This patch includes the fix for that as well. Review: https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close Reported by: Matt Jordan ASTERISK-23969 #close Reported by: Andrew Nagy ........ r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines Remove automerge properties :-( ........ r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines test_message: Fix strict-aliasing compilation issue ........ Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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dcf1ad14da |
Add module support level to ast_module_info structure. Print it in CLI "module show" .
ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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bb87796f67 |
ARI: Fix endpoint/channel subscription issues; allow for subscriptions to tech
This patch serves two purposes: (1) It fixes some bugs with endpoint subscriptions not reporting all of the channel events (2) It serves as the preliminary work needed for ASTERISK-23692, which allows for sending/receiving arbitrary out of call text messages through ARI in a technology agnostic fashion. The messaging functionality described on ASTERISK-23692 requires two things: (1) The ability to send/receive messages associated with an endpoint. This is relatively straight forwards with the endpoint core in Asterisk now. (2) The ability to send/receive messages associated with a technology and an arbitrary technology defined URI. This is less straight forward, as endpoints are formed from a tech + resource pair. We don't have a mechanism to note that a technology that *may* have endpoints exists. This patch provides such a mechanism, and fixes a few bugs along the way. The first major bug this patch fixes is the forwarding of channel messages to their respective endpoints. Prior to this patch, there were two problems: (1) Channel caching messages weren't forwarded. Thus, the endpoints missed most of the interesting bits (such as channel creation, destruction, state changes, etc.) (2) Channels weren't associated with their endpoint until after creation. This resulted in endpoints missing the channel creation message, which limited the usefulness of the subscription in the first place (a major use case being 'tell me when this endpoint has a channel'). Unfortunately, this meant another parameter to ast_channel_alloc. Since not all channel technologies support an ast_endpoint, this patch makes such a call optional and opts for a new function, ast_channel_alloc_with_endpoint. When endpoints are created, they will implicitly create a technology endpoint for their technology (if one does not already exist). A technology endpoint is special in that it has no state, cannot have channels created for it, cannot be created explicitly, and cannot be destroyed except on shutdown. It does, however, have all messages from other endpoints in its technology forwarded to it. Combined with the bug fixes, we now have Stasis messages being properly forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar), where bar has a single channel associated with it and foo has two channels associated with it. The messages would be forwarded as follows: channel PJSIP/foo-1 -- \ --> endpoint PJSIP/foo -- / \ channel PJSIP/foo-2 -- \ ---- > endpoint PJSIP / channel PJSIP/bar-1 -----> endpoint PJSIP/bar -- ARI, through the applications resource, can: - subscribe to endpoint:PJSIP/foo and get notifications for channels PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo - subscribe to endpoint:PJSIP/bar and get notifications for channels PJSIP/bar-1 and endpoint PJSIP/bar - subscribe to endpoint:PJSIP and get notifications for channels PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar Note that since endpoint PJSIP never changes, it never has events itself. It merely provides an aggregation point for all other endpoints in its technology (which in turn aggregate all channel messages associated with that endpoint). This patch also adds endpoints to res_xmpp and chan_motif, because the actual messaging work will need it (messaging without XMPP is just sad). Review: https://reviewboard.asterisk.org/r/3760/ ASTERISK-23692 ........ Merged revisions 419196 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419203 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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abd3e4040b |
Allow Asterisk to compile under GCC 4.10
This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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fbe0dfaf44 |
Fix dialplan function NULL channel safety issues
(closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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e4803bbd9e |
Voicemail: Remove mailbox identifier format (box@context) assumptions in the system.
This change is in preparation for external MWI support. Removed code from the system for normal mailbox handling that appends @default to the mailbox identifier if it does not have a context. The only exception is the legacy hasvoicemail users.conf option. The legacy option will only work for app_voicemail mailboxes. The system cannot make any assumptions about the format of the mailbox identifer used by app_voicemail. chan_sip and chan_dahdi/sig_pri had the most changes because they both tried to interpret the mailbox identifier. chan_sip just stored and compared the two components. chan_dahdi actually used the box information. The ISDN MWI support configuration options had to be reworked because chan_dahdi was parsing the box@context format to get the box number. As a result the mwi_vm_boxes chan_dahdi.conf option was added and is documented in the chan_dahdi.conf.sample file. Review: https://reviewboard.asterisk.org/r/3072/ ........ Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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4cdd02ce26 |
Reduce log level of a non-pubsub error message
Drop an error log message to debug level 1 since distributed device state functions correctly when receiving this message and it spams the logs. (closes issue ASTERISK-22410) Reported by: abelbeck Patches: asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903) asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903) ........ Merged revisions 401119 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401120 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401121 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401122 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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2de42c2a25 |
Multiple revisions 399887,400138,400178,400180-400181
........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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ad89dffb8a |
Prevent XMPP timeout on blank responses
Sometimes the Google Voice servers have a bad habit of sending out 1 byte replies to the xmpp resource. When a blank 1 byte reply is received from the socket the buffer attempts to wait (endlessly) for the rest of the reply from google which effectively blocks the socket and google voice calls will no longer come into the server. This patch allows the xmpp module to correctly detect empty packets and send out ping replies to google. It also sets a socket timeout on the default socket which prevents the xmpp socket from closing and preventing future google voice calls from coming into the server. Furthermore instead of sending an empty reply back to google we send a proper xmpp ping reply back. This also adds several more socket messages. (closes issue ASTERISK-22347) Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/2771 Patches: xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524) ........ Merged revisions 398618 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398619 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |