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2014-03-21res_pjsip_registrar.c: Miscellaneous cleanup in rx_task().Richard Mudgett
* Fix variable shadowing of 'updated' by renaming it to 'contact_update'. * Checked 'contact_update' for ast_sorcery_copy() failure. * Removed silly use of RAII_VAR() for 'contact_update'. ........ Merged revisions 410995 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21Make the AEL load process less chatty.Sean Bright
Switched a bunch of LOG_NOTICEs to ast_debug. This time without breaking the build. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21Revert r410981. aelparse blew up.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21Remove a LOG_NOTICE from ast_config_engine_register.Sean Bright
There is enough indication from the CLI that we are loading a realtime engine as it is. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21Make the AEL load process less chatty.Sean Bright
Switched a bunch of LOG_NOTICEs to ast_debug. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-20app_confbridge: Fix bug - users with startmuted set don't start mutedJonathan Rose
(closes issue ASTERISK-23461) Reported by: Chico Manobela Review: https://reviewboard.asterisk.org/r/3373/ ........ Merged revisions 410965 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410966 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-20assigned-uniqueids: Miscellaneous cleanup and fixes.Richard Mudgett
* Fix memory leak in ast_unreal_new_channels(). Made it generate the ;2 uniqueid on a stack variable instead of mallocing it. * Made send error response to ARI and AMI requests instead of just logging excessive uniqueid length and allowing truncation. action_originate() and ari_channels_handle_originate_with_id(). * Fixed minor truncating uniqueid hole when generating the ;2 uniqueid string length. Created public and internal lengths of uniqueid. The internal length can handle a max public uniqueid plus an appended ;2. * free() and ast_free() are NULL tolerant so they don't need a NULL test before calling. * Made use better struct initialization format instead of the position dependent initialization format. Also anything not explicitly initialized in the struct is initialized to zero by the compiler. * Made ast_channel_internal_set_fake_ids() use the safer ast_copy_string() instead of strncpy(). Review: https://reviewboard.asterisk.org/r/3371/ ........ Merged revisions 410949 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-19PJSIP: Allow for identify sections to be specified in sorcery.conf.Mark Michelson
"identify" is a special type of configuration object in PJSIP because unlike the other objects, it is not provided by the base res_pjsip module. Instead, it is provided by the res_pjsip_endpoint_identifier_ip module. If using the default sorcery wizard (config,criteria=type=identify) then things work because the module that applies the default wizard is the correct module. However, if attempting to use sorcery.conf to apply an alternate wizard, it was not possible. If you attempted to specify the identify object type in the res_pjsip section, then the object could not be registered since the object was undocumented for the res_pjsip module. There was no alternate configuration section defined for it, so you were out of luck if you wanted to override the default wizard. With this change, the identify section will properly have a sorcery.conf-based wizard applied when the identify definition is within the res_pjsip_endpoint_identifier_ip section. ........ Merged revisions 410933 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-19res_stasis: Fix a bug where the default bridge type was not set.Joshua Colp
........ Merged revisions 410918 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-19res_stasis: Extend bridge type to be a comma separated list of bridge ↵Joshua Colp
attributes. This change turns the bridge type field into a comma separated list of attributes. These attributes include: mixing, holding, dtmf_events, and proxy_media. By setting the various attributes a user can control the type of bridge created with the behavior they need for their application. (closes issue ASTERISK-23437) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3359/ ........ Merged revisions 410904 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-19res_ari: Fix documentation schema errorMatthew Jordan
........ Merged revisions 410890 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-18res_ari: Add notes about Asterisk HTTP server to the "enabled" config option ↵Rusty Newton
for the res_ari general section Added note and see-also reminding user to enable the HTTP server. (closes issue ASTERISK-22499) Reported by: Rusty Newton ........ Merged revisions 410876 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-18ARI: allow json content type with zero length bodyScott Griepentrog
When a request was received with a Content-type of json, the body was sent for json parsing - even if it was zero length. This resulted in ARI requests failing that were valid, such as a channel DELETE with no parameters. The code has now been changed to skip json parsing with zero content length. (closes issue SWP-6748) Reported by: Samuel Galarneau Review: https://reviewboard.asterisk.org/r/3360/ ........ Merged revisions 410858 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-18cdr: Add asserts for when we don't know about a CDR for a channelMatthew Jordan
In the CDR core, every channel should either be filtered out (due to being an 'internal' channel used as an implementation detail, such as playing media back into a bridge) or it should get a CDR. Even if that CDR ends up being discarded, we still give the channel a CDR in case we end up needing it. If we hit a situation where a channel does not have a CDR, we should blow up in -dev-mode. Asserts are appropriate for that. This patch adds those asserts, as they would have quickly caught the error fixed by r410814. ........ Merged revisions 410861 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-18res_pjsip: Fix memory leak of nameservers in off-nominal resolver creation ↵Joshua Colp
failure. Thanks Walter Doekes! ........ Merged revisions 410844 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-18res_fax_spandsp: Use g711_free() when available.Sean Bright
Per Johann Steinwendtner on the asterisk-dev mailing list: http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html g711_free() was introduced in spandsp 0.0.6pre4 and g711_release() became a noop. I opted not to remove the call to g711_release() since it is harmless and to call g711_free() if we have a sufficiently recent version of spandsp. (issue ASTERISK-20149) Reported by: Alexandr Gordeev ........ Merged revisions 410829 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410830 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-18stasis_cache: Use the right variable in the cache entry ao2 cmp function.Richard Mudgett
........ Merged revisions 410813 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-17res_pjsip: Enable PJSIP DNS client support.Joshua Colp
This change enables DNS client support within PJSIP. System nameservers are automatically discovered using res_init or res_ninit. If this fails then PJSIP will resort to using gethostbyname for resolution. By enabling this support we gain SRV support, failover, and weight support. (closes issue ASTERISK-23435) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3343/ ........ Merged revisions 410795 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-17res_pjsip_multihomed: Make address replacement less aggressive.Joshua Colp
This change makes the res_pjsip_multihomed module less aggressive when changing the address in messages. It will now only occur if the transport in use is bound to the any address OR if the system determined source address matches the bound address of the transport in use. Review: https://reviewboard.asterisk.org/r/3369/ ........ Merged revisions 410793 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-17callerid: Logic error in checksum processingRuss Meyerriecks
Callerid checksum-ing was being handled incorrectly here. When the checksum is calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This value will then fail the otherwise correct callerid message. This patch changes the logic to simply add the calculated checksum to the transmitted 2's compliment checksum. Review: https://reviewboard.asterisk.org/r/3356/ (closes issue ASTERISK-23488) ........ This is a merge of merged revisions 410750 410747 from http://svn.asterisk.org/svn/asterisk/branches/12 I didn't want a broken patch to be comitted to trunk so I pre-merge merged them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-17Revert changes to sorcery that accidentally got committed.Mark Michelson
These changes were still up for review and have not been approved yet. I must have had the changes in my working copy when making a different change. ........ Merged revisions 410696 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-17Fix stuck channel in ARI through the introduction of synchronous bridge actions.Mark Michelson
Playing back a file to a channel in an ARI bridge would attempt to wait until the playback concluded before returning. The method used involved signaling the waiting thread in the ARI custom playback function. The problem with this is that there were some corner cases that were not accounted for: * If a bridge channel could not be found, then we never would attempt the playback but would still attempt to wait for the playback to complete. * If the bridge playfile action failed to queue, we would still attempt to wait for the playback to complete. * If the bridge playfile action were queued but some circumstance caused the playback not to occur (the bridge dies, the channel is removed from the bridge), then we would never be notified. The solution to this is to move the waiting logic into the bridge code. A new bridge API function is added to queue a synchronous action on a bridge. The waiting thread is notified when the queued frame has been freed, either due to an error occurring or due to successful playback. As a failsafe, the waiting thread has a 10 minute timeout just in case there is a frame leak somewhere. Review: https://reviewboard.asterisk.org/r/3338 ........ Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-17app_confbridge: Add missing destructor call to announcer channel destructor.Richard Mudgett
........ Merged revisions 410671 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-16stasis/app.c: Add some extra debugging for subscription countsMatthew Jordan
Events are sent to a connected ARI application based on the things that ARI application cares about. These subscriptions can be set up implicitly - such as when that ARI application creates a new object - or explicitly, via the application resource's subscription operations. Debugging *why* something was being sent to an application - or why something was not being sent to an application - was a bit tricky, as there was no debug information for the subscriptions. This patch adds some debug level 3 statements that show the subscription counts for applications. (Level 3 was chosen as it matches the verbose level 3 statements elsewhere) ........ Merged revisions 410650 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-15framehook.h: Fix some doc typos.Russell Bryant
There were a number of instances in this header file where "function all" was intended to be "function call". This patch fixes that up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14Fix failing realtime sorcery tests.Mark Michelson
The store realtime callback needs to return a positive value for sorcery to treat the store as a success. ........ Merged revisions 410625 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14manager: fix memory leak in manager_add_filter functionJonathan Rose
(closes issue ASTERISK-23420) Reported by: Etienne Lessard Patches: manager_eventfilter_leak uploaded by Etienne Lessard (license 6394) ........ Merged revisions 410609 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410623 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14Remove an extra ast_cond_wait() that slipped through the patch.Mark Michelson
........ Merged revisions 410606 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410607 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14Handle the return values of realtime updates and stores more accurately.Mark Michelson
Realtime backends' update and store callbacks return the number of rows affected, or -1 if there was a failure. There were a couple of issues: * The config API was treating 0 as a successful return, and positive values as a failure. Now the config API treats anything >= 0 as a success. * res_sorcery_realtime was treating 0 as a successful return from the store procedure, and any positive values as a failure. Now sorcery treats anything > 0 as a success. It still considers 0 a "failure" since there is no change to report to observers. Review: https://reviewboard.asterisk.org/r/3341 ........ Merged revisions 410592 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14Prevent conflicts regarding unsolicited and solicited MWI to an endpoint.Mark Michelson
If an endpoint is receiving unsolicited MWI for a mailbox and then attempts to subscribe to an AOR that provides MWI for the same mailbox, then the SUBSCRIBE is rejected with a 500 response. Review: https://reviewboard.asterisk.org/r/3345 ........ Merged revisions 410590 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14uniqueid: Update CHANGES to reflect new featuresScott Griepentrog
Note the new features provided by uniqueid in the CHANGES file. (issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3316/ ........ Merged revisions 410588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14PJSIP: TOS values should be represented as decimals in sorcery objectsJonathan Rose
(closes issue ASTERISK-23235) Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/3324/ ........ Merged revisions 410574 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14Prevent delayed astdb syncs.Mark Michelson
The syncing thread sleeps for a second before waiting to be told to attempt to sync again. If a signal were sent during this sleeping period, we would end up having to wait until the next sync signal occurred in order to sync up the astdb. This code rearrangement also ensures that any pending transactions will be synced prior to Asterisk shutting down. Patches: db_sync.patch by John Hardin (License #6512) ........ Merged revisions 410556 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410559 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14ARI/bridges: Forward Playback/Recording Started/Finished to bridge topicJonathan Rose
(closes issue ASTERISK-23444) Reported by: Ben Merrills Review: https://reviewboard.asterisk.org/r/3340/ ........ Merged revisions 410558 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14res_mwi_external: Clear the stasis cache entry when the external MWI is deleted.Richard Mudgett
One of the things missing when external MWI support was added was the ability to clear the stasis cache entry of deleted external MWI mailboxes. Review: https://reviewboard.asterisk.org/r/3325/ ........ Merged revisions 410555 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-13cdr.c: Add missing aow_unlock(cdr) in off nominal path of handle_dial_message().Richard Mudgett
* Trivial common code hoisting in handle_bridge_leave_message(). * Some whitespace fixing. ........ Merged revisions 410541 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-13ARI: Ensure managing application receives ChannelEnteredBridge messagesKinsey Moore
This fixes an issue where a Stasis application running over ARI and subscribed to ari/events could miss the ChannelEnteredBridge event because it did not subscribe to the new bridge fast enough. To accomplish this, it subscribes the application controlling the channel to the new bridge before adding it to that bridge which required the stasis_app_control structure to maintain a reference to the stasis_app. (closes issue ASTERISK-23295) Review: https://reviewboard.asterisk.org/r/3336/ ........ Merged revisions 410527 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-13Multiple revisions 410509-410510Joshua Colp
........ r410509 | file | 2014-03-13 06:23:14 -0700 (Thu, 13 Mar 2014) | 2 lines res_pjsip_multihomed: Fix a bug where the 200 OK for a REGISTER would contain the wrong contact. ........ r410510 | file | 2014-03-13 06:24:17 -0700 (Thu, 13 Mar 2014) | 2 lines res_pjsip_multihomed: Remove change for testing fix. ........ Merged revisions 410509-410510 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-12res_musiconhold.c: Generate MOH start/stop events whenever the MOH stream is ↵Richard Mudgett
started/stopped. * Made res_musiconhold.c always post the MusicOnHoldStart/MusicOnHoldStop events when it actually starts/stops the music streams. This allows the events to always happen when MOH starts/stops. The event posting code was moved to the MOH alloc/release routines. * Made channel_do_masquerade() stop any MOH on the original channel before masquerading so the original channel will get a stop event with correct information. * Cleaned up a couple odd codings in moh_files_alloc() and moh_alloc() dealing with the music state variable. (issue ASTERISK-23311) Reported by: Benjamin Keith Ford Review: https://reviewboard.asterisk.org/r/3306/ ........ Merged revisions 410493 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-12app_confbridge: Make explicitly stop MOH if a user is kicked or hangs up ↵Richard Mudgett
while MOH is playing. When MOH is playing to a user in a conference and the user is kicked or hangs up from the conference then the AMI MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event: MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported by: Benjamin Keith Ford Review: https://reviewboard.asterisk.org/r/3306/ ........ Merged revisions 410490 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410491 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-12res_pjsip_multihomed: Fix a bug where outgoing messages for TCP would go out ↵Joshua Colp
using UDP. This change fixes a bug where the code which changes the transport did not check whether the message is going out over UDP or not before changing it. For TCP and TLS transports we don't need to change the transport as the correct one is already chosen. ........ Merged revisions 410471 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-11res_pjsip_multihomed: Add module which places the correct address within ↵Joshua Colp
messages. Due to how messages are handled within PJSIP it is not until a message is actually sent that the destination is reliably known. This means that the addresses placed within the message may not be of the interface the message is being sent out on. This module determines what interface a message is being sent on and updates the message to contain the correct address if applicable. This module was tested by myself in a virtualized environment with multiple interfaces and also by Kinsey Moore in the following configuration: Networks: * 10.24.16.0/21 ** hard phone ** default gateway * 10.24.64.0/21 ** softphone with pjsip-based stack Transport details: bind address: 0.0.0.0 protocol: UDP All endpoints were tested with explicitly configured transports and unconfigured transports. This was tested with inbound and outbound calls, both of which were experiencing detrimental effects from incorrect IP addresses in SIP messages. These effects were only experienced by the soft phone on the 10.24.64.0 network since the messages to the hard phone on the 10.24.16.0 network had the correct IP address. (closes issue ASTERISK-23020) Reported by: xrobau Review: https://reviewboard.asterisk.org/r/3102/ ........ Merged revisions 410451 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-10AST-2014-001: Stack overflow in HTTP processing of Cookie headers.Richard Mudgett
Sending a HTTP request that is handled by Asterisk with a large number of Cookie headers could overflow the stack. Another vulnerability along similar lines is any HTTP request with a ridiculous number of headers in the request could exhaust system memory. (closes issue ASTERISK-23340) Reported by: Lucas Molas, researcher at Programa STIC, Fundacion; and Dr. Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions 410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410381 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410383 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-10unqiueid: correct max uniqueid length testScott Griepentrog
This patch adds null string test prior to checking for a max uniqueid value that was added in r410157. ........ Merged revisions 410368 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-10AST-2014-002: chan_sip: Exit early on bad session timers requestKinsey Moore
This change allows chan_sip to avoid creation of the channel and consumption of associated file descriptors altogether if the inbound request is going to be rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 410308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410311 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410329 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-10AST-2014-003: res_pjsip: When handling 401/407 responses don't assume a ↵Joshua Colp
request will have an endpoint. This change removes the assumption that an outgoing request will always have an endpoint and makes the authenticate_qualify option work once again. (closes issue ASTERISK-23210) Reported by: Joshua Colp ........ Merged revisions 410306 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-08pjsip_cli: Create pjsip show channel and contact, and general cli code cleanup.George Joseph
Created the 'pjsip show channel' and 'pjsip show contact' commands. Refactored out the hated ast_hashtab. Replaced with ao2_container. Cleaned up function naming. Internal only, no public name changes. Cleaned up whitespace and brace formatting in cli code. Changed some NULL checking from "if"s to ast_asserts. Fixed some register/unregister ordering to reduce deadlock potential. Fixed ast_sip_location_add_contact where the 'name' buffer was too short. Fixed some self-assignment issues in res_pjsip_outbound_registration. (closes issue ASTERISK-23276) Review: http://reviewboard.asterisk.org/r/3283/ ........ Merged revisions 410287 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-08resource_channels: Check if a passed in ID is NULL before checking its lengthMatthew Jordan
Calling strlen on a NULL string is explosive. This patch checks whether or not the passed in string is NULL or zero length before checking to see if the string is too long. ........ Merged revisions 410274 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07chan_sip: Fix deadlock of monlock between unload_module and do_monitorCorey Farrell
Release monlock before calling pthread_join. This ensures do_monitor cannot freeze by locking monlock during module unload. (closes issue ASTERISK-21406) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3284/ ........ Merged revisions 410224 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410225 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410226 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07sorcery: correct field register argument listScott Griepentrog
This fixes mistakes I previously made in merging gtjoseph's changes with mine. ........ Merged revisions 410211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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