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2013-10-22res_parking: Give parking timeout comebacktoorigin channel DTMF features.Richard Mudgett
Parking timeouts did not set any DTMF features for the channel calling the parker back. * Added code to set the parkedcalltransfers, parkedcallreparking, parkedcallhangup, and parkedcallrecording options appropriately for the channels when a parking timeout occurs. The recall channel DTMF options are set using the BRIDGE_FEATURES channel variable to allow the other timeout options to have the DTMF features available. (closes issue ASTERISK-22630) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2942/ ........ Merged revisions 401422 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22res_parking: Update XML documention for DTMF features after parking timeout.Richard Mudgett
* Updated the XML documentation to indicate that the parkedcalltransfers, parkedcallreparking, parkedcallhangup, and parkedcallrecording configuration options also apply to parking timeouts. (issue ASTERISK-22630) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2942/ ........ Merged revisions 401420 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-21Fixed malformed Access-Control-Allow-Methods header. Was causing Safari to ↵David M. Lee
barf on POST and DELETE. ........ Merged revisions 401106 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-19Return a channel snapshot when originating using ARI, and subscribe the ↵Joshua Colp
Stasis application to it. This change allows a user of ARI to know what channel it has originated and also follow any progress. If a Stasis application is provided it will be automatically subscribed to the originated channel immediately. (closes issue ASTERISK-22485) Reported by: David Lee Review: https://reviewboard.asterisk.org/r/2910/ ........ Merged revisions 401281 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18res_parking: Remove setting useless flag.Richard Mudgett
........ Merged revisions 401271 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Add channel lock protection around translation path setup.Richard Mudgett
Most callers of ast_channel_make_compatible() happen before the channels enter a two party bridge. With the new bridging framework, two party bridging technologies may also call ast_channel_make_compatible() when there is more than one thread involved with the two channels. * Added channel lock protection in set_format() and ast_channel_make_compatible_helper() when dealing with the channel's native formats while setting up a translation path. * Fixed best_src_fmt and best_dst_fmt usage consistency in ast_channel_make_compatible_helper(). The call to ast_translator_best_choice() got them backwards. * Updated some callers of ast_channel_make_compatible() and the function documentation. There is actually a difference between the two channels passed in. * Fixed the deadlock potential in res_fax.c dealing with ast_channel_make_compatible(). The deadlock potential was already there anyway because res_fax called ast_channel_make_compatible() with chan locked. (closes issue ASTERISK-22542) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2915/ ........ Merged revisions 401239 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-17res_parking: Fix bug where reloading immediately wipes new parkpos extensionsJonathan Rose
(closes issue ASTERISK-22631) Reported by: Kevin Harwell ........ Merged revisions 401158 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-17Reduce log level of a non-pubsub error messageKinsey Moore
Drop an error log message to debug level 1 since distributed device state functions correctly when receiving this message and it spams the logs. (closes issue ASTERISK-22410) Reported by: abelbeck Patches: asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903) asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903) ........ Merged revisions 401119 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401120 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401121 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16ARI: Fix crash when POST /playback/{id}/control does not have an operation ↵Richard Mudgett
parameter. (closes issue ASTERISK-22680) Reported by: John Bigelow ........ Merged revisions 401107 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16Clarify documentation for channel and bridge listKinsey Moore
This makes it clear that the ARI API calls for listing channels and bridges will list all channels or bridges in the system and not just those that are in or are controlled by a Stasis application. (closes issue ASTERISK-22635) Reported by: Kevin Harwell ........ Merged revisions 401087 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16Use POST / DELETE to toggle ARI bridge mohPaul Belanger
Review: https://reviewboard.asterisk.org/r/2911/ ........ Merged revisions 401040 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15Ensure bridge record error responses validateKinsey Moore
This adds the list of expected errors to the /bridges/{bridgeId}/record ARI documentation so that outbound 4xx errors validate properly. Previously, this would result in a response validation failure. (closes issue ASTERISK-22627) Reported by: Joshua Colp ........ Merged revisions 401018 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15Use POST / DELETE to toggle hold / moh for ARI channelsPaul Belanger
This change updates how we handle toggle events, rather then create two different function names, we'll just use POST / DELETE from HTTP to handle it. Review: https://reviewboard.asterisk.org/r/2906/ ........ Merged revisions 400999 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-14pjsip outbound registration: Log message says received a 408 when we didn'tKevin Harwell
If the server didn't exist that we are trying to register to the log message would say that a 408 was received from that server when in reality one wasn't. Added log messages stating no response was received if the response does not exist. (closes issue ASTERISK-22554) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2893/ ........ Merged revisions 400890 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-14Remove duplicate module info blockMatthew Jordan
The module info block was repeated twice. Once is sufficient. ........ Merged revisions 400881 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-13Fix a race condition in res_pjsip_session with rapidly terminating the session.Joshua Colp
The INVITE session state callback wrongly assumes that a session will always exist, but when rapidly terminating the session this assumption goes out the window. As all handler code for the INVITE session state callback requires the session it will now just exit immediately if no session exists. (closes issue ASTERISK-22668) Reported by: John Bigelow ........ Merged revisions 400872 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-12Fix realm comparison for outbound authKinsey Moore
When generating the list of authentication credentials to pass to PJSIP, Asterisk was using the raw pointer of a pj_str_t which is not always NULL-terminated. This sometimes resulted in incorrect text for the realm and a failure to match the realm for authentication purposes which was causing the outbound nominal auth pjsip basic call test to bounce. This now uses the pj_str_t that contains the realm instead of generating a new one. Thanks to John Bigelow for helping to narrow this down. ........ Merged revisions 400863 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-11Multiple revisions 400508,400842-400843,400848David M. Lee
........ r400508 | dlee | 2013-10-03 23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line Corrected response class for stopPlayback ........ r400842 | dlee | 2013-10-10 14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line Correct some ARI wiki rendering errors ........ r400843 | dlee | 2013-10-10 14:26:19 -0500 (Thu, 10 Oct 2013) | 1 line Updated /play resource docs. The playback of http: resources isn't implemented... yet ........ r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5 lines Fix a stupid copy/paste error in ARI docs. Patches: ari-doc-patch.txt uploaded by jbigelow (license 5091) ........ Merged revisions 400508,400842-400843,400848 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-10Perform validation of permanent contacts on AORs in res_pjsip.Joshua Colp
........ Merged revisions 400833 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-10Fix an assertion in res_pjsip when specifying an invalid outbound proxy.Joshua Colp
This change fixes two issues when setting an outbound proxy: 1. The outbound proxy URI was not parsed and validated during configuration. 2. If an outgoing dialog was created and the outbound proxy could not be set an assertion would occur because the usage count on the dialog was not decremented. The documentation has also been updated to specify that a full URI must be specified for the outbound proxy. (closes issue ASTERISK-22672) Reported by: Antti Yrjola ........ Merged revisions 400824 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-09Use 'z' as the format specifier for size_tMatthew Jordan
Using 'lu' will produce a compiler warning for some versions of gcc and on some architectures. 'z' should be portable as a format specifier for size_t. ........ Merged revisions 400812 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-08Add PJSIP_HEADER function for manipulation of SIP headers in the PJSIP stackMatthew Jordan
This patch adds support to the PJSIP stack in Asterisk for SIP header manipulation. Note that this is analagous to SIPAddHeader/SIPRemoveHeader. For PJSIP_HEADER, an incoming supplemental session callback is registered that takes the pjsip_hdrs from the incoming session and stores them in a linked list in the session datastore. Calls to PJSIP_HEADER traverse over the list and return the nth matching header where 'n' is the 'number' argument to the function. When adding a header, the first call creates a datastore and linked list and adds the datastore to the session. The header is then created as a pjsip_hdr and added to the list. An outgoing supplemental session callback then traverses the list and adds the headers to the outgoing pjsip_msg. When removing a header, the list created with PJSIP_HEADER(add,...) is traversed and all matching entries are removed. (closes issue ASTERISK-22498) Reported by: George Joseph patch: res_pjsip_header_funcs_v1.patch uploaded by george.joseph (License 6322) ........ Merged revisions 400771 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-08Switch from using pjsip_strerror to pj_strerror.Mark Michelson
pjsip_strerror is only aware of PJSIP-specific error codes. pj_strerror() is aware of all PJProject error codes and OS-specific error codes. This specifically fixes an oft-seen error in transport configuration code where EADDRINUSE would result in "Unknown PJSIP error 120098" instead of a useful message. ........ Merged revisions 400749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-08Fix STUN crash when using IPv6 any addressKinsey Moore
Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4 candidates are also added. (closes issue ASTERISK-21917) Reported by: Torrey Searle Patches: 0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334) ........ Merged revisions 400681 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400682 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-08Push CLI qualify into the threadpool.Mark Michelson
If you run Asterisk in the background and then connect to it through a separate console, the thread that runs CLI commands is not registered with PJLIB. Thus PJLIB does not like it when you attempt to send OPTIONS requests from that thread. So now we push the task into the threadpool, which we know to be registered with PJLIB. Thanks to Antti Yrjola for reporting this. ........ Merged revisions 400680 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-08Make app_queue and res_agi independent of AMI being enabled.Richard Mudgett
The https://reviewboard.asterisk.org/r/2888/ review changes manager to not subscribe to stasis when it is disabled for performance reasons. When manager is disabled app_queue and res_agi decline to load and fail to clean up what they have already allocated. * Made app_queue and res_agi clean up allocated resources when they decline to load. * Made app_queue and res_agi use their own subscriptions to the stasis topics instead of borrowing manager's message router structure inappropriately. (closes issue ASTERISK-22604) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/2902/ ........ Merged revisions 400671 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-04chan_pjsip: Make logger togglable without loading/unloadingJonathan Rose
This patch makes the res_pjsip_logger do a few things... First, it will be built and installed by default now, so end users won't need to enable it in menuselect. Second, while it is loaded, it no longer will immediately issue log messages. Upon loading, it is in the disabled state and must be turned on with the new CLI command. The CLI command 'pjsip set logger <on/off/host> has been added and can be used to do the following: pjsip set logger on: Enables logger for all PJSIP traffic pjsip set logger off: Disables logger for all PJSIP traffic pjsip set logger host <host>: Enables logger for the specific host Review: https://reviewboard.asterisk.org/r/2900/ ........ Merged revisions 400542 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-04ARI: Add subscription supportMatthew Jordan
This patch adds an /applications API to ARI, allowing explicit management of Stasis applications. * GET /applications - list current applications * GET /applications/{applicationName} - get details of a specific application * POST /applications/{applicationName}/subscription - explicitly subscribe to a channel, bridge or endpoint * DELETE /applications/{applicationName}/subscription - explicitly unsubscribe from a channel, bridge or endpoint Subscriptions work by a reference counting mechanism: if you subscript to an event source X number of times, you must unsubscribe X number of times to stop receiveing events for that event source. Review: https://reviewboard.asterisk.org/r/2862 (issue ASTERISK-22451) Reported by: Matt Jordan ........ Merged revisions 400522 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-04Enclose the To URI and update its user portion if a request user has been ↵Joshua Colp
specified. ........ Merged revisions 400520 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-04Replace the connection address at the SDP level if altering the SDP with the ↵Joshua Colp
external media address. ........ Merged revisions 400510 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03Fix a crash caused by muting and unmuting a channel in ARI without ↵Joshua Colp
specifying a direction. (closes issue ASTERISK-22637) Reported by: Scott Griepentrog Patch by Matt Jordan, whose office I have taken over in the name of Canada. ........ Merged revisions 400401 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03res_rtp_multicast: Ensure SSRC is set properlyKinsey Moore
This fixes a bug where the SSRC field on multicast RTP can be stuck at 0 which can cause problems for endpoints trying to make sense of incoming streams. (closes issue ASTERISK-22567) Reported by: Simone Camporeale Patches: 22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536) ........ Merged revisions 400393 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400394 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400395 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03Cache string values of formats on ast_format_cap() to save processing.Mark Michelson
Channel snapshots have string representations of the channel's native formats. Prior to this change, the format strings were re-created on ever channel snapshot creation. Since channel native formats rarely change, this was very wasteful. Now, string representations of formats may optionally be stored on the ast_format_cap for cases where string representations may be requested frequently. When formats are altered, the string cache is marked as invalid. When strings are requested, the cache validity is checked. If the cache is valid, then the cached strings are copied. If the cache is invalid, then the string cache is rebuilt and copied, and the cache is marked as being valid again. Review: https://reviewboard.asterisk.org/r/2879 ........ Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03Fix crashes in res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected ↵Joshua Colp
and external_media_address is set. The callback function for changing the media address in streams wrongly assumes that a connection line will always be present. This is false as no line is present if a stream has been rejected. (closes issue ASTERISK-22645) Reported by: Rusty Newton ........ Merged revisions 400360 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02Multiple revisions 400318-400319Mark Michelson
........ r400318 | mmichelson | 2013-10-02 17:08:49 -0500 (Wed, 02 Oct 2013) | 12 lines Remove unnecessary waits from stasis. Since caches are updated on publisher threads, there is no need to wait for the cache updates to occur after a stasis message is published. In the case of chan_pjsip device state changes, this set of changes caused an improvement to performance. Review: https://reviewboard.asterisk.org/r/2890 ........ r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed, 02 Oct 2013) | 3 lines Remove svn:mergeinfo property. ........ Merged revisions 400318-400319 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02Fix a crash in res_pjsip_t38 caused by the wrong assumption that a session ↵Joshua Colp
will always have a channel. When starting up or shutting down this assumption is false. ........ Merged revisions 400284 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02MALLOC_DEBUG: Fix some misuses of free() when MALLOC_DEBUG is enabled.Richard Mudgett
* There were several places in ARI where an external library was mallocing memory that must always be released with free(). When MALLOC_DEBUG is enabled, free() is redirected to the MALLOC_DEBUG version. Since the external library call still uses the normal malloc(), MALLOC_DEBUG complains that the freed memory block is not registered and will not free it. These cases must use ast_std_free(). * Changed calls to asprintf() and vasprintf() to the equivalent ast_asprintf() and ast_vasprintf() versions respectively. ........ Merged revisions 400270 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02Fix a random one way audio issue in PJSIP.Joshua Colp
Due to the asynchronous design of the PJMEDIA SDP negotiator it was possible for the SDP to be negotiated *after* a channel was created and after it was being wait on by an application. It is only after negotiation occurs that the file descriptors for RTP are placed on the channel. Since the channel was already being waited on these file descriptors were not monitored, causing incoming media to never be read. This change wakes up any application waiting on the channel so that added file descriptors end up being monitored. (closes issue AST-1227) Reported by: John Bigelow ........ Merged revisions 400256 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02Allow specifying a channel to dial an extension and context in an ARI dial ↵Joshua Colp
operation. (issue ASTERISK-22625) Reported by: Scott Griepentrog ........ Merged revisions 400254 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02Retrieve and store the hostname only once so multiple threads do not ↵Joshua Colp
potentially initialize it at the same time. ........ Merged revisions 400245 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-30Multiple revisions 399887,400138,400178,400180-400181David M. Lee
........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-30chan_sip: Allow Asterisk to retry after 403 on registerKinsey Moore
This adds a global option in chan_sip to allow it to continue attempting registration if a 403 is received, clearing the cached nonce and treating it as a non-fatal response. Normally, this would cause registration attempts to that endpoint to stop. This also adds a similar per-outbound-registration option to chan_pjsip which allows the retry interval to be altered for 403 responses to REGISTER requests. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874/ Reported by: Rudi ........ Merged revisions 400137 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400140 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400141 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-28res_pjsip_notify: Add documentationMatthew Jordan
We forgot to add documentation for res_pjsip_notify, which would prevent it from being loaded. Whoops. This patch also updates res_pjsip_notify to use pjsip_notify.conf, which now has its own sample file in the configs directory as well. Review: https://reviewboard.asterisk.org/r/2835/ ........ Merged revisions 400121 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-28res_rtp_asterisk: Correct erroneous lost packet information in RTCP reportsMatthew Jordan
RTCP's calculation of the number of lost packets in an RTP stream is based on that stream's sequence number count, the number of received packets, and how many packets we expect to receive. When the SSRC for an RTP stream changes, there can - and almost always will be - a large jump in the next packet's timestamp and sequence number. If we don't reset the number of received packets, sequence number count, and other metrics used by RTCP, the next RR/SR report will use the previous SSRC's values to calculate the lost packet count for the new SSRC - resulting in a very large number of lost packets. This patch modifies res_rtp_asterisk such that, if it detects a SSRC change, it will reset the various values used by the RTCP calculations. From the perspective of RTCP, this appears as a new media stream - which is what it is. Review: https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174) Reported by: Thomas Arimont ........ Merged revisions 400089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400093 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400108 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-27app_cdr and res_parking: Fix some resource leaks.Richard Mudgett
* app_cdr left the ResetCDR application registered. * res_parking leaked a ref to config global. (closes issue ASTERISK-22566) Reported by: Corey Farrell Patches: ASTERISK-22566-r2.patch (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 400020 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-27res_pjsip: crash when using localnet and external_signaling_address optionsKevin Harwell
There was a collision of mod_data use on the transaction between using a nat hook and an session response callback. During state change it was assumed what was in the mod_data was nothing or the response callback. However, it was possible for it to also contain a nat hook thus resulting in a bad cast and a crash. Added the ability to store multiple data elements in mod_data via a hash table. In this instance, mod_data now stores a hash table of the two values that can be retrieved using an associated string key. (closes issue ASTERISK-22394) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2843/ ........ Merged revisions 399990 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-26pjsip: race condition in registrarKevin Harwell
While handling a registration request a race condition could occur if/when two+ clients registered at the same time. This happened when one request obtained a copy of the current contacts for an AOR and another request did the same before the first request updated. Thus the second would update and overwrite the first (or vice-versa depending on which actually updated first). In the case of it being the same contact two "add" events would be raised. pjsip registration handling is now serialized to alleviate this issue. (closes issue AST-1213) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2860/ ........ Merged revisions 399897 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-25Broke the build - Fixing XML DTD violation added in r399782, missing <para> ↵Rusty Newton
tags inside a <note> ........ Merged revisions 399798 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-25Fixing documentation for the configOption "external_media_address" of both ↵Rusty Newton
Endpoints and Transports Re-using some of Mark Michelson's text from an E-mail discussion for: * Modifying synopsis for both options * Adding description to both options * Changing name of "external_media_address" for Endpoint configuration to "media_address" in anticipation of the option name being changed. (As it is not really specific to external destinations) (issue ASTERISK-22405) (closes issue ASTERISK-22405) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2850/ ........ Merged revisions 399781 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-23Fix crash in res_pjsip on load if error occurs, and prevent unloading of ↵Joshua Colp
res_pjsip and res_pjsip_session. During load time in res_pjsip if an error occurred the operation would attempt to rollback all operations done during load. This is not permitted by PJSIP as it will assert if the operation has not been done. This fix changes the code so it will only rollback what has been initialized already. Further changes also prevent res_pjsip and res_pjsip_session from being unloaded. This is due to limitations within PJSIP itself. The library environment can only be changed to a certain extent and does not provide the ability, currently, to deinitialize certain required functionality. (closes issue ASTERISK-22474) Reported by: Corey Farrell ........ Merged revisions 399624 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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