asterisk

Commit Graph

Author	SHA1	Message	Date
Scott Griepentrog	ba811ae1c3	stun: correct attribute string padding to match rfc When sending the USERNAME attribute in an RTP STUN response, the implementation in append_attr_string passed the actual length, instead of padding it up to a multiple of four bytes as required by the RFC 3489. This change adds separate variables for the string and padded attributed lengths, and performs padding correctly. Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/4139/ ........ Merged revisions 427874 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427875 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427876 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2d9471ab1f	Fix race condition that could result in ARI transfer messages not being sent. From reviewboard: "During blind transfer testing, it was noticed that tests were failing occasionally because the ARI blind transfer event was not being sent. After investigating, I detected a race condition in the blind transfer code. When blind transferring a single channel, the actual transfer operation (i.e. removing the transferee from the bridge and directing them to the proper dialplan location) is queued onto the transferee bridge channel. After queuing the transfer operation, the blind transfer Stasis message is published. At the time of publication, snapshots of the channels and bridge involved are created. The ARI subscriber to the blind transfer Stasis message then attempts to determine if the bridge or any of the involved channels are subscribed to by ARI applications. If so, then the blind transfer message is sent to the applications. The way that the ARI blind transfer message handler works is to first see if the transferer channel is subscribed to. If not, then iterate over all the channel IDs in the bridge snapshot and determine if any of those are subscribed to. In the test we were running, the lone transferee channel was subscribed to, so an ARI event should have been sent to our application. Occasionally, though, the bridge snapshot did not have any channels IDs on it at all. Why? The problem is that since the blind transfer operation is handled by a separate thread, it is possible that the transfer will have completed and the channels removed from the bridge before we publish the blind transfer Stasis message. Since the blind transfer has completed, the bridge on which the transfer occurred no longer has any channels on it, so the resulting bridge snapshot has no channels on it. Through investigation of the code, I found that attended transfers can have this issue too for the case where a transferee is transferred to an application." The fix employed here is to decouple the creation of snapshots for the transfer messages from the publication of the transfer messages. This way, snapshots can be created to reflect what they are at the time of the transfer operation. Review: https://reviewboard.asterisk.org/r/4135 ........ Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427870 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cc4c396647	main/rtp_engine: Fix crash when processing more than one RTCP report info block Asterisk - in res_rtp_asterisk - only understands a single RTCP report info block. When the RTCP information was refactored in the RTP Engine to be pushed over the Stasis message bus, I put in the hooks into the engine to handle multiple RTCP report info blocks, in the hope that a future RTP implementation would be able to provide that data. Unfortunately, res_rtp_asterisk has a tendency to "lie": (1) It will send RTCP reports with a reception_report_count greater than 1 (which is pulled directly from the RTCP packet itself, so that part is correct) (2) It will only provide a single report block When the rtp_engine goes to convert this to a JSON blob, hilarity ensues as it looks for a report block that doesn't exist. This patch updates the rtp_engine to be a bit more skeptical about what it is presented with. While this could also be fixed in res_rtp_asterisk, this patch prefers to fix it in the engine for two reasons: (1) The engine is designed to work with multiple RTP implementation, and hence having it be more robust is a good thing (tm) (2) res_rtp_asterisk's handling of RTCP information is "fun". It should report the correct reception_report_count; ideally it should also be giving us all of the blocks - but it is definitely not designed to do that. Going down that road is a non-trivial effort. Review: https://reviewboard.asterisk.org/r/4158/ ASTERISK-24489 #close Reported by: Gregory Malsack Tested by: Gregory Malsack ASTERISK-24498 #close Reported by: Beppo Mazzucato Tested by: Beppo Maazucato ........ Merged revisions 427762 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427763 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ec1a7654f3	Fix leak in AMI Action Bridge Add missing reference cleanup for newly created bridge. ASTERISK-24281 Reported by: Stefan Engström Review: https://reviewboard.asterisk.org/r/4154/ ........ Merged revisions 427736 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427737 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	dbb8f0a935	pbx: Fix off-nominal case where a freed extension may still be used. If during the operation of adding an extension a priority is added but fails it is possible for the extension to be freed but still exist in the PBX core. If this occurs subsequent lookups may try to access the extension and end up in freed memory. This change removes the extension from the PBX core when the priority addition fails and then frees the extension. ASTERISK-24444 #close Reported by: Leandro Dardini Review: https://reviewboard.asterisk.org/r/4162/ ........ Merged revisions 427709 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427710 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427711 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	21c41e4542	manager: Fix HTTP connection reference leaks. Fix reference leak that happens if (session && !blastaway). ASTERISK-24505 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4153/ ........ Merged revisions 427641 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427642 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427643 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a68baad74f	Bridge DTMF hooks: Made audio pass from the bridge while waiting for more matching digits. * Made collecting DTMF digits for the DTMF feature hooks pass frames from the bridge. * Made collecting DTMF digits possible by other bridge hooks if there is a need. ASTERISK-24447 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4123/ ........ Merged revisions 427493 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427494 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	4d80f223af	main/file.c: fix possible extra ast_module_unref to format modules. fn_wrapper only adds a reference to the format's module if the file was able to be opened. If not this causes an unmatched ast_module_unref in filestream_destructor. Move ast_module_ref to get_stream. ASTERISK-24492 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4149/ ........ Merged revisions 427464 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427465 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427466 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	7e2369310c	Fix unintential memory retention in stringfields. * Fix missing / unreachable calls to __ast_string_field_release_active. * Reset pool->used to zero when the current pool->active reaches zero. ASTERISK-24307 #close Reported by: Etienne Lessard Tested by: ibercom, Etienne Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........ Merged revisions 427380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427381 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427382 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d5de94201e	config: Make text_file_save and 'dialplan save' escape semicolons in values. When a config file is read, an unescaped semicolon signals comments which are stripped from the value before it's stored. Escaped semicolons are then unescaped and become part of the value. Both of these behaviors are normal and expected. When the config is serialized either by 'dialplan save' or AMI/UpdateConfig however, the now unescaped semicolons are written as-is. If you actually reload the file just saved, the unescaped semicolons are now treated as start of comments. Since true comments are stripped on read, any semicolons in ast_variable.value must have been escaped originally. This patch re-escapes semicolons in ast_variable.values before they're written to file either by 'dialplan save' or config/ast_config_text_file_save which is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting issues nearby in pbx_config.c Tested-by: George Joseph ASTERISK-20127 #close Review: https://reviewboard.asterisk.org/r/4132/ ........ Merged revisions 427275 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427276 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	bd42a09d7f	config: BUG: Restore ability for non-templ to be used as base objs in config. My recent refactor of config.c accidentally removed the capability for an object to inherit from a non-template object. This patch restores the capability to inherit from both template and non-template objects. Tested-by: George Joseph Reported-by: Scott Griepentrog ASTERISK-24487 #close Review: https://reviewboard.asterisk.org/r/4147/ ........ Merged revisions 427227 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427228 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	bdc35c77b9	Fix crash caused by merge error on review 4138 When merging from 12 to 13 there were conflicts, I mistakenly had the loop run ast_closestream(others[0]) when it should be ast_closestream(others[x]). ........ Merged revisions 427181 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	509c04ef38	Fix ast_writestream leaks Fix cleanup in __ast_play_and_record where others[x] may be leaked. This was caught where prepend != NULL && outmsg != NULL, once realfile[x] == NULL any further others[x] would be leaked. A cleanup block was also added for prepend != NULL && outmsg == NULL. 11+: Fix leak of ast_writestream recording_fs in app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/ ........ Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427024 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427025 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427026 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	85c1822a9d	func_jitterbuffer: fix frame leaks. Fix code paths where it is possible for frames to leak. Fix uninitialized variable in jb_get_fixed and jb_get_adaptive. ASTERISK-22409 #related Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4128/ ........ Merged revisions 427019 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427020 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427021 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ced81afff2	audiohooks: Clean references to formats Cleanup references to in_translate[x].format and out_translate[x].format in ast_audiohook_detach_list. ASTERISK-24465 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4124/ ........ Merged revisions 426803 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	67e496c275	manager: Unsubscribe from acl_change_sub at shutdown. ASTERISK-24453 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4110/ ........ Merged revisions 426524 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426525 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8e9f593e3a	main/bridge: Destroy features struct on off nominal path during bridge impart When a channel is imparted to a bridge, the invocation of the function may provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart, the caller must assume that ownership has passed to the function, as in all paths the function destroys the struct prior to returning (as its purpose is to configure the behavior of the channel while in the bridge). On one off nominal path - where the channel already has a PBX thread - the struct was not being destroyed. This patch fixes that glitch. ASTERISK-24437 #close Reported by: Scott Griepentrog ........ Merged revisions 426431 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426432 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f4b4d42630	main/manager: Fix typo in AMI event documentation of "OriginateResponse" The parameter name is "Response", not "Resonse". ASTERISK-24430 #close Reported by: Dafi Ni ........ Merged revisions 426366 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426367 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426368 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c084728690	Documentation: Improve documentation for ExtensionStatus AMI events Review: https://reviewboard.asterisk.org/r/4085/ ........ Merged revisions 426120 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2165868be7	translage.c: Fix regression when generating translation path strings. Fix the AMI Status action read and write translation path strings from growing for each channel in the status event list by reseting the ast string given to ast_translate_path_to_str() to fill in the given translation path. ........ Merged revisions 426079 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	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
Richard Mudgett	f91cb1207c	AMI: Add missing VarSet events when a channel inherits variables. There should be AMI VarSet events when channel variables are inherited by an outgoing channel. Also local;2 should generate VarSet events when it gets all of its channel variables from channel local;1. ASTERISK-24415 #close Reported by: Richard Mudgett Patches: jira_asterisk_24415_v12.patch (license #5621) patch uploaded by Richard Mudgett Review: https://reviewboard.asterisk.org/r/4074/ ........ Merged revisions 425782 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425783 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	873d956144	main/cdr: Use 'time' when rescheduling batched CDRs as opposed to 'size' When refactoring CDRs to use the configuration framework, a 'whoops' was introduced where the CDR batch size was used when rescheduling a batch, as opposed to the time duration. This patch corrects that obvious mistake. ASTERISK-24426 #close Reported by: Shane Blaser ........ Merged revisions 425735 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425736 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c2ec5f0f6f	config: Fix inf loop using ast_category_browse and ast_variable_retrieve Fix infinite loop when calling ast_variable_retrieve inside an ast_category_browse loop when there is more than 1 category with the same name. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4089/ ........ Merged revisions 425713 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425714 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	104fca5001	config: Fix SEGV in unit test with MALLOC_DEBUG With MALLOC_DEBUG the /main/config config_basic_ops test was causing a SEGV while doing an ast_category_delete in an ast_category_browse loop. Apparently this never worked but was also never tested. I removed the test, added 2 notes to config.h indicating that it's not supported and added a few lines of code to ast_category_delete to prevent the SEGV should someone attempt it in the future. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4078/ ........ Merged revisions 425525 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425526 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	87b5006ff0	Scheduler: Fix a nasty scheduler caching bug which makes new tasks not execute Tasks that were marked for pending deletion in the scheduler would be moved to the cache for later reuse, but after being recycled the deleted mark wouldn't be removed resulting in fresh tasks being deleted without reason... and immediately moved back into the cache where they could be reused again. This could cause horrendous things to happen in just about anything that used a scheduler. ASTERISK-24321 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4071/ ........ Merged revisions 425503 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425504 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c61b66e107	stasis_channels.c: Resolve unfinished Dials when doing masquerades (Part 2) Masquerades into and out of channels that are involved in a dial operation don't create the expected dial end event. The missing dial end event goes against the model for things like CDRs and generating Dial end manager actions and such. There are four cases: 1) A channel masquerades into the caller channel. The case happens when performing a blonde transfer using the channel driver's protocol. 2) A channel masquerades into a callee channel. The case happens when performing a directed call pickup. 3) The caller channel masquerades out of dial. The case happens when using the Bridge application on the caller channel. 4) A callee channel masquerades out of dial. The case happens when using the Bridge application on a peer channel. As it turned out, all four cases need to be handled instead of just the first one. ASTERISK-24237 Reported by: Richard Mudgett ASTERISK-24394 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/ ........ Merged revisions 425430 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425455 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c7e6b6ba3d	manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	98d5b7090d	bridge: During a smart bridge operation provide a more complete bridge to the old technology. When a smart bridge operation occurs and a bridge transitions from one technology to another the old technology is provided the channels formerly in it and told that they are leaving. Unfortunately the bridge provided along with them is incomplete. The bridge, despite there being channels in it, contains none. This forces technology implementations to have additional logic when channels are leaving or to store their own duplicated state. This change makes the bridge more complete so it contains the expected channels. Now that the bridge is complete special logic within bridge_native_rtp is no longer needed and has been removed. Review: https://reviewboard.asterisk.org/r/4057/ ........ Merged revisions 425242 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	37b5f52da7	CallerID: Fix parsing regression This fixes a regression in callerid parsing introduced when another bug was fixed. This bug occurred when the name was composed entirely of DTMF keys and quoted without a number section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/ ........ Merged revisions 425152 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425153 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425154 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425155 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	cc595f7353	res_phoneprov: Refactor phoneprov to allow pluggable config providers This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424964 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0f50e8856b	cdr.c: Make turning on CDR debug a one step process instead of two. Now "cdr set debug on" doesn't also require "core set verbose 1" to see CDR debug output. ........ Merged revisions 424941 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424942 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	5e50638539	Stasis: Relegate log message to dev-mode This error message primarily applies to development tasks and will now only show up when dev-mode is enabled via configure. ........ Merged revisions 424850 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	3dfc485e35	Indexer: Format message types may not exist In Asterisk 13+, any given message type is not guaranteed to exist even if Asterisk comes up correctly since creation of the message type could be declined. The indexer should not prevent Asterisk from starting under these conditions. ........ Merged revisions 424833 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	d8bbf1ec1d	Stasis: Only log errors for non-declined types When message type creation is declined via stasis.conf, certain operations log errors assuming that the declined type is being used before initialization or after destruction. These error messages get quite spammy for oft used message types and should not be logged in the first place since the message type is validly NULL. Reported by: Matt DiMeo ........ Merged revisions 424769 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	f7225da08a	data: Properly access formats in capabilities structure when adding codecs. Formats within a capabilities structure are addressed starting at 0, not 1. Assuming 1 causes it to exceed an array. ASTERISK-24389 #close Reported by: Kevin Harwell ........ Merged revisions 424752 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424753 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	39bd5b7a70	message: Don't close an AMI connection on SendMessage action error If SendMessage encounters an error (such as incorrect input provided to the action), it will currently return -1. Actions should only return -1 if the connection to the AMI client should be closed. In this case, SendMessage causing the client to disconnect is inappropriate. This patch causes the action to return 0, which simply causes the action to fail. Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354 #close Reported by: Peter Katzmann patches: sendMessage.patch uploaded by Peter Katzmann (License 5968) ........ Merged revisions 424690 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424691 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c384532aa4	features.c: Fix lingering channel ref while Bridge() application is active. Using the Bridge application to bridge a channel that is executing an applicaiton such as Wait results in a lingering Surrogate channel in the CLI "core show channels" output even though it has already hungup. * Fix bridge_exec() to not hold onto the current_dest_chan ref once it has been put into the bridge. * Eliminated bridge_exec()'s use of RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged revisions 424668 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424669 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	3a87f32dc0	sdp_srtp: Add new lines to some WARNING messages ........ Merged revisions 424646 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424647 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	9611ef4f1e	Release AMI connections on shutdown. ASTERISK-24378 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4037/ ........ Merged revisions 424578 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424579 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424580 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424581 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	70301b0438	audiohooks: Reevaluate the bridge technology when an audiohook is added or removed. Adding a mixmonitor to a channel causes the bridge to change technologies from native to simple_bridge so the call can be recorded. However, when the mixmonitor is stopped the bridge does not switch back to the native technology. * Added unbridge requests to reevaluate the bridge when a channel audiohook is removed. * Moved the unbridge request into ast_audiohook_attach() ensure that the bridge reevaluates whenever an audiohook is attached. This simplified the mixmonitor and chan_spy start code as well. * Added defensive code to stop_mixmonitor_full() in case additional arguments are ever added to the StopMixMonitor application. * Made ast_framehook_detach() not do an unbridge request if the framehook does not exist. * Made ast_framehook_list_fixup() do an unbridge request if there are any framehooks. Also simplified the loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4046/ ........ Merged revisions 424506 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424507 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0165c5f95a	chan_pjsip: Fix deadlock when masquerading PJSIP channels. Performing a directed call pickup resulted in a deadlock when PJSIP channels were involved. A masquerade needs to hold onto the channel locks while it swaps channel information between the two channels involved in the masquerade. With PJSIP channels, the fixup routine needed to push a fixup task onto the PJSIP channel's serializer. Unfortunately, if the serializer was also processing a task that needed to lock the channel, you get deadlock. * Added a new control frame that is used to notify the channels that a masquerade is about to start and when it has completed. * Added the ability to query taskprocessors if the current thread is the taskprocessor thread. * Added the ability to suspend/unsuspend the PJSIP serializer thread so a masquerade could fixup the PJSIP channel without using the serializer. ASTERISK-24356 #close Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/4034/ ........ Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424472 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	4967478d18	sorcery: Prevent SEGV in sorcery_wizard_create when there's no create function When you call ast_sorcery_create() you don't necessarily know which wizard is going to be invoked. If it happens to be a wizard like 'config' that doesn't have a 'create' virtual function you get a segfault in the sorcery_wizard_create callback. This patch catches the null function pointer, does an ast_assert, and logs an error. Review: https://reviewboard.asterisk.org/r/4044/ ........ Merged revisions 424447 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424448 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	4246652603	Manager: Add missing fields and documentation for CoreShowChannels This corrects some issues introduced in the responses to the CoreShowChannels AMI command as well as adding documentation for the responses. The command in Asterisk 12 was missing the following fields: Duration, Application, ApplicationData, and BridgedChannel and BridgedUniqueID (replaced with BridgeId). ASTERISK-24262 #close Reported by: Mitch Claborn Review: https://reviewboard.asterisk.org/r/4040/ ........ Merged revisions 424423 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424424 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9d2bc0675a	threadpool.c: Minor cleanup fixes. * Fix threadpool_alloc() prototype. * Add missing off-nominal NULL check of pool in threadpool_alloc(). * searializer_create() does not need to create the object with a lock as the lock is not used. ........ Merged revisions 424096 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424097 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	e0abb82ab8	core: Ouch, forgot to undo a test free() in r423978. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	37179a2b1f	core: Don't allow free to mean ast_free (and malloc, etc..). This gets rid of most old libc free/malloc/realloc and replaces them with ast_free and friends. When compiling with MALLOC_DEBUG you'll notice it when you're mistakenly using one of the libc variants. For the legacy cases you can define WRAP_LIBC_MALLOC before including asterisk.h. Even better would be if the errors were also enabled when compiling without MALLOC_DEBUG, but that's a slightly more invasive header file change. Those compiling addons/format_mp3 will need to rerun ./contrib/scripts/get_mp3_source.sh. ASTERISK-24348 #related Review: https://reviewboard.asterisk.org/r/4015/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	63a4da4a0d	cli.c: Fix tab completion "module load" when MALLOC_DEBUG is enabled. r421600 conflicted with r155763. ASTERISK-24348 #close ........ Merged revisions 423657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423658 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423659 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423660 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	64a9e5f001	main/channel: Unlock channel in off-nominal path In r423414 (13) / r423415 (trunk), an API call that determines if a format capability structure is empty was added. This returns true if the format capability structure is completely empty or "none". A check for this was added in channel.c's set_format call. Unfortunately, when this check was true, it returned from the function while still holding the channel lock. This caused the CDR unit tests - which have a tendency to create channels with no formats - to deadlock. Whoops. This patch unlocks the channel on the off-nominal path. ........ Merged revisions 423641 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	6dae345674	Stasis_channels: Resolve unfinished Dials when doing masquerades Masquerades into channels that are in the dialing state don't end their dial and this goes against the model for things like CDRs and generating Dial end manager actions and such. ASTERISK-24237 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3990/ ........ Merged revisions 423525 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423530 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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4725 Commits (ba811ae1c3f80cb0d691fbe0dbdac2db1d4d960a)