asterisk

Commit Graph

Author	SHA1	Message	Date
David M. Lee	965777ccfc	Various fixes for OS X This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sigset() functions. Fixed some format strings for portability. * Fixed compilation issues with res_timing_kqueue (although tests still fail on OS X). * Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by: George Joseph ASTERISK-24544 #close Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/4327/ ........ Merged revisions 431092 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	a8ae5a7bcb	dynamic realtime: Updates fail to work due to update fields being passed over When a crash was fixed due to usage of the REALTIME function in r423003, a regression was introduced into ast_update2_realtime where the update fields passed to the function would be skipped and the lookup field processed twice. The use of this function is a bit interesting: A variable argument list is used with two sentinel values - the first marks the end of the lookup fields/values; the second marks the end of the update fields/values. Unfortunately, ast_update2_realtime parses over the lookup fields twice, as opposed to parsing over the update fields. This causes the lookups to succeed, but the updates itself to have no effect. Note that the most common instance of this problem occurred in app_voicemail during the updating of a mailbox password. Thanks to the issue reporter, Paddy Grice, for pointing out the problem. Review: https://reviewboard.asterisk.org/r/4356/ ASTERISK-24231 ASTERISK-24626 #close Reported by: Paddy Grice ........ Merged revisions 431072 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	b69b0d12ee	app_confbridge: Shorten CBRec channel names to CBRec/<conf_name>-<seq-num> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c780223507	app_confbridge: Make CBRec channel names more unique. Channel names should be different from other channels in the system while the channel exists. * Use a sequence number for CBRec channels instead of a random number because the same random number could be picked again for the next CBRec channel. ........ Merged revisions 431052 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b38be992b1	app_confbridge: Whitespace Because there is sometimes no sence to any whitespace. ........ Merged revisions 431049 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431050 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
David M. Lee	89610adda5	Add depend on pjproject to res_pjsip_config_wizard.c ........ Merged revisions 431030 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	ca02121ef7	Investigate and fix memory leaks in Asterisk Fixed memory leaks that were found in Asterisk. ASTERISK-24693 #close Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/4347/ ........ Merged revisions 430999 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	49cbfa7de6	Fix typo's (retrieve, specified, address). ........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430998 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	874cb5615d	chan_sip: Case insensitive comparison of "defaultuser" parameter. All the other configuration options are case insensitive, so this one should be too. ASTERISK-24355 #close Reported by: HZMI8gkCvPpom0tM patches: ast.patch uploaded by HZMI8gkCvPpom0tM (License 6658) ........ Merged revisions 430993 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430994 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9bff4eeca3	Bridge core: Pass a ref with the swap channel when joining a bridge. When code imparts a channel into a bridge to swap with another channel, a ref needs to be held on the swap channel to ensure that it cannot dissapear before finding it in the bridge. * The ast_bridge_join() swap channel parameter now always steals a ref for the swap channel. This is the only change to the bridge framework's public API semantics. * bridge_channel_internal_join() now requires the bridge_channel->swap channel to pass in a ref. ASTERISK-24649 Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4354/ ........ Merged revisions 430975 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e67ca431ee	res_pjsip_outbound_registration.c: Minor code cleanup. * Add an allocation failure check and assert in sip_outbound_registration_response_cb(). * Made sip_outbound_registration_state_destroy() handle partially created state objects from sip_outbound_registration_state_alloc(). Review: https://reviewboard.asterisk.org/r/4366/ ........ Merged revisions 430957 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	49f405fe4c	stasis transfer: fix a race condition on stasis bridge push After a bridge transfer completes where a local replacement channel is used, a stasis transfer message with the details of the transfer is sent. This is processed by stasis which then sets the stasis app name and replaced channel snapshot on the replacement channel. However, since a separate thread was already started to run stasis on the new replacement channel, a race was on to see if the message processing would be completed before the app name was needed, otherwise the channel would be hung up. This change moves the calls used to set the stasis app name and the replace snapshot to the bridge_stasis_push function callback from the bridge transfer logic, allowing the steps to be completed earlier and more deterministically, and the race elimianted. NOTE: the swap channel parameter to bridge_stasis_push (and thus all bridge push callbacks) must always be present when performing a swap with another channel. ASTERISK-24649 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4341/ ........ Merged revisions 430939 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	7fcc9ce8bc	apps/app_voicemail: Trigger MWI notification with MixMonitor m() option The MixMonitor m() option allows a recording to be pushed to a specific voicemail mailbox. If the message is delivered to the mailbox's INBOX, however, no MWI notification is currently raised. This patch corrects the issue by properly calling notify_new_state from the msg_create_from_file function. This will cause MWI to be triggered if the message was placed in the mailbox's INBOX. ASTERISK-24709 #close Reported by: Gareth Palmer patches: app_voicemail-430919.patch uploaded by Gareth Palmer (License 5169) ........ Merged revisions 430920 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430921 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	38738a7316	res_pjsip_outbound_registration.c: Move unref to a better place. Move an unconditional unref of client_state so it doesn't look like it could be used after the last ref has destroyed it. ........ Merged revisions 430902 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5835bf7a7f	channels/chan_sip: Fix registration leak during reload When the SIP registrations were migrated to using ao2 in what was then trunk, the explicit destruction of the registrations on module reload was removed and not replaced with an ao2 equivalent. Debugging done by Stefan Engström, the issue reporter, on ASTERISK-24673 confirmed that the reference in the registry_list container was being leaked. Since the purpose of cleanup_all_regs is to prep a registration for destruction, this function now calls an ao2_callback function callback with the OBJ_MULTIPLE \| OBJ_NODATA \| OBJ_UNLINK flags used to remove the registrations. This cleans up each registration, and also removes it from the registration container registry_list. Review: https://reviewboard.asterisk.org/r/4355/ ASTERISK-24640 #close Reported by: Max Man ASTERISK-24673 #close Reported by: Stefan Engström Tested by: Stefan Engström ........ Merged revisions 430864 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	958a41a884	AMI: Add documentation for the missing Cdr/CEL events. This patch adds AMI event documentation for the Cdr and CEL AMI events. Note that while these events do share fields with each other and with other channel related events, they do not contain all of the fields in a standard channel snapshot, nor is the description of the fields identical. As such, the patch opts for documentation for each field, for each event. Review: https://reviewboard.asterisk.org/r/4350/ ASTERISK-24671 #close Reported by: Dan Jenkins ........ Merged revisions 430862 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	4740ef50f4	apps/app_dial: Don't publish DialEnd twice on unexpected GoSub/Macro values The Dial application has some interesting options with the mid-call Macro (M) and GoSub (U) options. If the MACRO_RESULT/GOSUB_RESULT returns specific values, the Dial application will take some action upon the channels involved in the dial operation (such as hanging up a particular party, etc.) The Dial application ensures that a Stasis message is published in the event that MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial operation, so that there is a corresponding DialEnd event published in AMI/ARI for the DialBegin event that preceeded it. A bug exists where that same DialEnd event will be published on Stasis even if the value returned in MACRO_RESULT/GOSUB_RESULT is not one that the Dial application cares about. This causes two DialEnd events to be published - one with the MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is all sorts of wrong. This patch fixes the bug by ensuring that we only publish a DialEnd message to Stasis if the Dial application's mid-call Macro/GoSub returns something that Dial cares about. Review: https://reviewboard.asterisk.org/r/4336 ASTERISK-24682 #close Reported by: Matt Jordan ........ Merged revisions 430842 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	228fdb3f4e	main/rtp_engine: Format NTP timestamps as unsigned longs When the RTCP reports are created, the NTP timestamps are stored as strings, as JSON does not have an integer type long enough to store the value. However, on 32-bit systems, a signed long may overflow for some portion of the timestamp. This patch corrects the overflow by formatting the timestamps as unsigned longs. ........ Merged revisions 430840 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Ashley Sanders	804ab70f9d	ARI: Fixed crash that occurred when updating a bridge when the optional query parameter 'name' was not supplied. Prior to this changeset, posting to the: /ari/bridges/{bridgeId} endpoint without specifying a value for the [name] query parameter, would crash Asterisk if the bridge you are attempting to create (or update) had the same ID as an existing bridge. The internal mechanism of the POST operation interpreted a null value for name, thus resulting in an error condition that crashed Asterisk. ASTERISK-24560 #close Reported By: Kinsey Moore Review: https://reviewboard.asterisk.org/r/4349/ ........ Merged revisions 430818 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e4738a59eb	CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge. Calling ast_channel_bridge_peer() cannot be done while holding any channel locks. The reported issue hit the deadlock in chan_iax2, but an audit of the ast_channel_bridge_peer() calls found three more locations where the same deadlock can occur. * Made CHANNEL(peer), res_fax, and the SNMP agent not call ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I had to rework the logic to not hold the channel lock. * Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done for legacy reasons that no longer apply. * Removed the iax.conf forcejitterbuffer option. It is now always enabled when the jitterbuffer option is enabled. If you put a jitter buffer on a channel it will be on the channel. ASTERISK-24600 #close Reported by: Jeff Collell Review: https://reviewboard.asterisk.org/r/4342/ ........ Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	14b8e03dad	contrib/scripts/install_prereq: Don't install 32-bit packages on 64-bit hosts On Debian based systems, the install_prereq tool uses a search command on Debian that results in selecting both 64-bit and 32-bit packages. Besides the waste of disk space, this can actually cause aptitude use 100% of memory on a VM with 1GB of RAM as it tried to work out all of the 32-bit package dependencies. This patch filters out the 32-bit packages on a 64-bit machine, and leaves 32-bit machines alone. ASTERISK-24048 #close Reported by: Ben Klang Tested by: Ben Klang, Matt Jordan patches: install_prereq_64-bit_compat.patch uploaded by Ben Klang (License 5876) ........ Merged revisions 430798 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430799 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	112bf1597e	app_voicemail: Temp message left after review/hangup with ODBC/IMAP backend When using ODBC or IMAP storage, temporary files created on the file system must be disposed of using the DISPOSE macro. The DELETE macro will map to a deletion function for the backend storage, but does not clean up any local files created as a result of the operation. When using voicemail with the operator and review options enabled, pressing 0 to enter the menu, followed by 1 to save the message, followed by any other DTMF press to delete the message, will result in the temporary file lingering on the file system. This patch properly calls DISPOSE after the DELETE. This causes the local file to be disposed of. ASTERISK-24288 #close Reported by: LEI FU patches: voicemail_odbc_review_fix.diff uploaded by LEI FU (License 6640) ........ Merged revisions 430795 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430796 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	7dc784ffa9	Call extension state callbacks at hint creation. When a hint gets created, any subsequent device or presence state changes result in extension status events getting sent out to interested parties. However, at the time of hint creation, no such event gets sent out, so watchers of extension state are potentially left in the dark until the first state change after hint creation. Patch contributed by John Hardin (License #6512) ........ Merged revisions 430776 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e43912f3f3	res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions. The first thing this patch fixes is UAS dialogs. Previously if a transport was configured on an endpoint and an inbound session was created there was no guarantee that requests sent on the dialog would use the correct transport and address information. This has now been fixed so an explicitly configured transport is taken into account. The second thing this patch fixes is res_pjsip_multihomed. The res_pjsip_multihomed module attempts to determine what transport a message should go out on and what addressing information should go into the message itself. In a scenario where multiple transports exist bound to the same IP address but a different port the code would incorrectly alter the transport and change the message to the wrong transport. This change makes the res_pjsip_multihomed module smarter so it will only change the transport and address information in the message when it is possible and makes sense. ASTERISK-24615 #close Reported by: David Justl Review: https://reviewboard.asterisk.org/r/4331/ ........ Merged revisions 430755 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	07e2a48ab1	REVERTING res_pjsip: make it unloadable Due to the original patch causing memory corruptions the patch is being removed until the problem can be resolved. ........ Merged revisions 430734 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	1111944afb	Change PJProject version requirement for ca_list_path transport option in CHANGES file. ........ Merged revisions 430716 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	831acba826	Fix problem where a hung channel could occur on a failed blind transfer. Different clients react differently to being told that a blind transfer has failed. Some will simply send a BYE and be done with it. Others will attempt to reinvite themselves back onto the call. In the latter case, we were creating a new channel and then leaving it to sit forever doing nothing. With this code change, that new channel will not be created and the dialog with the transferring channel will be cleaned up properly. ASTERISK-24624 #close Reported by Zane Conkle Review: https://reviewboard.asterisk.org/r/4339 ........ Merged revisions 430714 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	023fa0f9e8	Add support for the ca_list_path option for PJSIP transports. This allows for a path to be specified that has a collection of CA certificates in it. ASTERISK-24575 #close Reported by cloos Patches: pj-ca-path-trunk.diff uploaded by cloos (License #5956) Review: https://reviewboard.asterisk.org/r/4344 ........ Merged revisions 430709 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a8ea2f9287	res_fax.c, res_fax_spandsp.c: Remove redundant locking. When FAX was developed, apparently the faxregistry.container used to be a linked list that was converted to an ao2 container. Some of the replacement ao2 container operations still had explicit lock/unlocks around them. Three off nominal code paths in res_fax.c and res_fax_spandsp.c unlock the channel even though the routine did not lock the channel and other code paths in the routine do not unlock the channel. Review: https://reviewboard.asterisk.org/r/4340/ ........ Merged revisions 430687 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9b1c36d3fa	res_fax.c, res_fax_spandsp.c: Fix some curlies on the end of function definitions. ........ Merged revisions 430685 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430686 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	1e605d950b	res_pjsip_outbound_registration: Fix race condition when reloading and listing registrations. Due to the split of outbound registration state from configuration it is possible during a reload for a "pjsip show registrations" CLI command to be executed which gets an older snapshot of the configuration. This configuration may include outbound registrations which have been removed due to a reload operation occurring at the same time. The code for printing the outbound registration did not take this into account but now it does. AST-1506 #close Review: https://reviewboard.asterisk.org/r/4338/ ........ Merged revisions 430664 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f11fb76205	configure: If cross-compiling, assume we have working semaphores The Asterisk 13 configure.ac checks for HAS_WORKING_SEMAPHORE but does not have an option for cross-compiling so it fails with an exit. Since we're cross- compiling, we can't exactly go looking for the header. The semaphore.h header is relatively common: * It's part of the POSIX standard * It's part of GNU C Library As such, we assume that it will be present when cross-compiling. As such, this patch defaults "HAS_WORKING_SEMAPHORE" to "1" if cross-compiling is detected. If you're cross-compiling to a platform that doesn't support this, then make sure you re-define this to 0. ASTERISK-24663 #close Reported by: abelbeck patches: asterisk-13-anonymous-semaphores.patch uploaded by abelbeck (License 5903) ........ Merged revisions 430646 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	49542a794b	res_pjsip: make it unloadable The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4311/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) ........ Merged revisions 430628 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	67234b3ee2	Prevent slow graceful shutdown when outbound publications never started. The code was missing the case for explicitly destroying an outbound publication when Asterisk had never actually published anything. The result was that Asterisk would hang for a while on a graceful shutdown. With this change, the case is taken into account, and on a graceful shutdown, these publications are destroyed without the need to actually send a PUBLISH request. ASTERISK-24655 #close Reported by Kevin Harwell Review: https://reviewboard.asterisk.org/r/4325 ........ Merged revisions 430608 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	3eec8e4c44	build_tools/mkpkgconfig: Fix Cflags concatenation error in asterisk.pc The mkpkgconfig script incorrectly concatenates Cflags options together. As an example, the following: Cflags: -I/usr/include/libxml2 -g3 Is instead generated as: Cflags: -I/usr/include/libxml2-g3 This patch corrects the generation of Cflags in mkpkgconfig such that the Cflags options are output correctly. Review: https://reviewboard.asterisk.org/r/3707/ ASTERISK-23991 #close Reported by: Diederik de Groot patches: fix_mkpkgconfig.diff uploaded by Diederik de Groot (License 6600) ........ Merged revisions 430589 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430590 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	1780de95e4	app_macro: Don't restore the calling location on a channel redirect. v11: If a channel redirect to a macro exten of a macro that is active happens, the redirect location doesn't get executed. Instead the original macro location is restored and gets reexecuted. v13: An additional effect happens if a parked call times out to an extension in the macro that parked the call then the macro is reexecuted instead of the expected park return location. * Made not restore the macro calling location on an AST_SOFTHANGUP_ASYNCGOTO. * Increased the locked channel range when setting up the macro execution environment to cover things that should be done while the channel is locked. * Removed unnecessary NULL tests before calling ast_free() in _macro_exec(). ASTERISK-23850 #close Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4292/ ........ Merged revisions 430564 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430565 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0e631a541d	chan_pjsip: Add configure check for 'pjsip_get_dest_info' function. The 'pjsip_get_dest_info' function is used to determine if the signaling transport of the dialog is secure or not. This function was added in PJSIP 2.3 and does not exist in earlier versions. This configure check allows Asterisk to build and run with older versions at the loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of this argument will require upgrading to PJSIP 2.3. ASTERISK-24665 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4329/ ........ Merged revisions 430546 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4dd6b6ff59	AMI: Revert non-backwards compatible changes from earlier commit. * Reverted the change to astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. Unfortunately changing the case of a returned value is not a backward compatible change so for now FAXSessions is going to have to remain inconsistent with all of the other AMI list actions. * Reverted the minor protocol error fix in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". Caught by the testsuite. ASTERISK-24049 ........ Merged revisions 430528 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	aa7e06f797	configs/samples/features.conf.sample: Document attended transfer DTMF options The sample config was missing the configuration options for DTMF attended transfer completion scenarios. The configuration options 'atxferabort', 'atxfercomplete', 'atxferthreeway', and 'atxferswap' are now documented in the appropriate configuration file. ASTERISK-24678 #close Reported by: Niklas Larsson patches: features.conf.sample.diff uploaded by Niklas Larsson (License 5068) ........ Merged revisions 430526 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c7ea108e02	Revert -r430452 It needs to be redone for the next major AMI version change instead. ASTERISK-24049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9065488ddd	main/syslog: Allow dynamic logs, such as security events, to log to the syslog The security event log uses a dynamic log level (SECURITY) that is registered with the Asterisk logging core. Unfortunately, the syslog would ignore log statements that had a dynamic log level associated with them. Because the syslog cannot handle ad hoc dynamic log levels, this patch treats any dynamic log entries sent to the syslog as logs with a level of NOTICE. ASTERISK-20744 #close Reported by: Michael Keuter Tested by: Michael L. Young, Jacek Konieczny patches: asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 430506 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430507 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b38acbce6e	funcs/func_curl: Fix memory leak when CURLOPT channel datastore is destroyed When the channel datastore associated with the usage of CURLOPT on a specific channel is freed, the underlying structure holding the list of options is not disposed of. This patch properly frees the structure in the datastore .destroy callback. ASTERISK-24672 #close Reported by: Kristian Hogh patches: func_curl-memory-leak.diff uploaded by Kristian Hogh (License 6639) ........ Merged revisions 430487 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430488 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	fba836cc02	sip_to_pjsip: improve ability to parse input files General improvements to SIP to PJSIP conversion utility: 1) track default section of input file to allow parsing an include file that doesn't specify a [section] 2) informatively handle case of assignment without [section] 3) correctly handle getting sections from included files - [section]'s are inherited by included file 4) provide null string as default transport bind ip 5) gracefully handle missing portions of registration string 6) denote steps of operation during conversion and confirm top level files as a convenience ASTERISK-24474 #close Review: https://reviewboard.asterisk.org/r/4280/ Reported by: John Kiniston ........ Merged revisions 430469 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	5b30938394	app_bridge: return to the next dialplan priority When app_bridge grabs a channel and puts it into a bridge, the channel should then continue where it left off in the dialplan after the bridge has ended. Although it stores the current dialplan location as an after bridge goto on the channel, it was executing the same priority again instead of going to the next priority. By swapping the "specific" version of bridge_set_after_goto with bridge_set_after_go_on, the next priority in the dialplan is executed instead. ASTERISK-24637 #close Review: https://reviewboard.asterisk.org/r/4322/ Reported by: John Bigelow ........ Merged revisions 430467 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ef34a05f21	AMI: Remove no longer used parameter from astman_send_listack(). Follow-up issue to -r430435 from reviewboard review. ASTERISK-24049 Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	52a7cdb101	AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	77ee23210d	res_fax: Add T.38 negotiation timeout option This change makes the T.38 negotiation timeout configurable via 't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously hard coded to be 5000 milliseconds. This change also handles T.38 switch failures by aborting the fax since in the case where this can happen, both sides have agreed to switch to T.38 and Asterisk is unable to do so. Review: https://reviewboard.asterisk.org/r/4320/ ........ Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430416 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	8786fe13a4	res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdown If you do a 'core (shutdown\|restart) graceful' persistent subscriptions won't survive. If you do a 'core (shutdown\|restart) now' or asterisk terminates for some reason, they do. Here's why... When asterisk shuts down gracefully, it sends a 'NOTIFY/terminated' to subscribers for each subscription. This not only tells the subscribers that the dialog/state machine is done, it also frees the last reference to the subscription tree which causes the persistent subscription to get deleted from astdb. When asterisk restarts, nothing's left. Just preventing the delete from astdb doesn't work because we already told the subscriber to terminate the dialog so we can't restart it even if it was still in astdb. Everything works OK if asterisk terminates unexpectedly because we never send the 'terminated' message so on restart, the subscription is still in astdb and the subscriber is none the wiser. This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for persistent connections. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4318/ ........ Merged revisions 430397 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430398 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c55f86c69d	res_pjsip_outbound_registration: Fix reference leak. Every time a registration started, sip_outbound_registration_response_cb bumps the ref count on client_state then pushes a handle_registration_response task. handle_registration_response never unreffed it though. So every time a registration goes out, the ref count goes up by one. This patch adds the unreffs to handle_registration_response. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4303/ ........ Merged revisions 430395 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	030facce94	res_pjsip_outbound_registration: Fix several reload issues There are 2 issues with reloading registrations... 1. The 'can_reuse_registration' test wasn't considering the intervals or expiration in its determination of whether a registration changed or not so if you changed any of the intervals or the expiration and reloaded, the object would get reloaded but the actual timers wouldn't change. can_reuse_registration now does a sorcery diff on the old and new objects instead of discretely testing certain fields. Now if you change expiration for instance, and reload, the timer is updated and re-registration will occur on the new value. 2. If you mung up your password on an outbound registration you get a permanent failure. If you fix the password (on the outbound_auth object) and reload, nothing tells outbound_registration to try again because the registration itself didn't change. This patch adds an observer on the "auth" object type and if any auth changes, existing registration states are searched and those in a REJECTED_PERMANENT state are retried. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4304/ ........ Merged revisions 430373 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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