asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	3ccb6a9ea5	Fix the CDR CLI command 'cdr show active {channel}' When the switch from channel names to channel unique IDs happened, the poor CLI command got left in the dust. This fixes the command so that users can once again see how Asterisk is messing up your billing information. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400286 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	94a673a878	Fix a crash in res_pjsip_t38 caused by the wrong assumption that a session will always have a channel. When starting up or shutting down this assumption is false. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tzafrir Cohen	de4ce10095	man pages for astdb2bdb and astdb2sqlite3 Review: https://reviewboard.asterisk.org/r/2898/ ........ Merged revisions 400279 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1a7783c7f8	MALLOC_DEBUG: Fix some misuses of free() when MALLOC_DEBUG is enabled. * 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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c114871a68	sig_ss7: Fix compiler warnings. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	1dd63fbdfa	Reduce channel snapshot creation and publishing by up to 50%. This change introduces the ability to stage channel snapshot creation and publishing by suppressing the implicit creation and publishing that some functions have. Once all operations are executed the staging is marked as done and a single snapshot is created and published. Review: https://reviewboard.asterisk.org/r/2889/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	1e22b90efe	Fix a random one way audio issue in PJSIP. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	add6d9a012	Allow specifying a channel to dial an extension and context in an ARI dial operation. (issue ASTERISK-22625) Reported by: Scott Griepentrog git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	48132c1d8c	Retrieve and store the hostname only once so multiple threads do not potentially initialize it at the same time. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	dfc077bdf6	chan_dahdi: Fix analog parking using flash-hook. Transferring an analog call using a flash-hook to parking would fail to park the call and result in an invalid ao2 object unref. * Park the correct bridged channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	d3f1f5369f	Features: Rearm the parking config options have moved warning for each reload. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	2efa0948dc	Filter out internal channels for bridge leave messages and parked call messages Granted, if you manage to park a Conference announcer channel, something has gone horrifically wrong. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	7b9d250ffa	configuration samples: Pull all parking related stuff out of features.conf This patch also adds documentation for parking from features.conf to res_parking.conf git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	dafa9e4e72	Parse arguments passed to the CDR_PROP function correctly I can only blame this on a bad merge, because this in no way worked properly the way it was written. Mea culpa. The function should now parse its arguments correctly and function properly. (Note that the API used by the CDR_PROP function has working unit tests... this was merely bad coding of the actual registered function) (closes issue ASTERISK-22613) Reported by: Private Name git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	fa0d221bd3	Remove spurious event raised when CDRs are reloaded The Reload event is now raised by the module loading core. As such, the Reload event in the CDR engine was a duplicate and not needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	516dbe86a0	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	9d21631aee	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	e88afe2022	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	52d73f05f4	chan_sip: Allow Asterisk to retry after 403 on register 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	48d90d947d	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	2438636a31	res_pjsip_notify: Add documentation 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	efa6676548	res_rtp_asterisk: Correct erroneous lost packet information in RTCP reports 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	07da7acd63	Add check for openSUSE when detecting bfd library In ASTERISK-17842, some additional library checks were added to the configure script so that the bfd library could be found on CentOS and Fedora systems. As it turns out, openSUSE requires an additional library. This patch adds another check to the configure script for openSUSE that will add that library. Review: https://reviewboard.asterisk.org/r/2885/ (closes issue AST-1169) Reported by: Guenther Kelleter ........ Merged revisions 400073 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400075 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	0b97ac8e87	CDR: Improve handling of parking; resolve assertion when originating into park This patch covers two problems: 1) Currently, when a call is transferred into a parking lot from a bridge (using either the blind transfer or one touch parking mechanisms), the application fails to be set to "Park" in the resulting CDR record for the parked channel. This is due to the ParkedCall message arriving before the BridgeEnter for the channel entering the parking bridge. The ParkedCall message isn't handled as the CDR for the channel has already been finalized (due to the channel having left its two party bridge), and the BridgeEnter - which creates the new CDR - doesn't have the parking information. This patch modifies the behavior so that reception of a ParkedCall message will - if not handled by a CDR chain - cause a new CDR to be created and put into the Parking state. 2) It fixes a FRACK that occurred when a channel is originated into a parking space. The DialedPending state - which occurs for both Dialed and Originated channels - assumed that it couldn't handle the parking transitions due to it having a Party B; however, Originated channels don't have a Party B. As such, the existing CDR needs to transition into the parking state - this patch does that. Review: https://reviewboard.asterisk.org/r/2877/ (closes issue ASTERISK-22482) Reported by: Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	d4bce93552	app_queue: Make manager events tolerant of Local channel shenanigans app_queue currently attempts to handle Local channel optimizations in an effort to provide accurate information in Stasis messages (and their corresponding AMI events) as well as the Queue log. Sometimes, however, things don't go as planned. Consider the following scenario: SIP/foo <-> L;1 <-> L;2 <-> SIP/agent SIP/agent answers, triggering a Local channel optimization. app_queue will normally do the following: * Listen for the Local optimization events and update our agent accordingly to SIP/agent in the queue log and messages * When we get a hangup, publish the AgentComplete event based on our information (SIP/foo and SIP/agent) However, as with all things that depend on sanity from something as capricious as Local channels, things can go wrong: (1) SIP/agent immediately hangs up upon answering. This triggers a race condition between termination messages coming from SIP/agent and the ongoing Local channel optimization messages. (Note that this can also occur with SIP/foo) (2) In a race condition, Asterisk can (rarely) deliver the hangup messages prior to the Local channel optimization. In that case, the messages may arrive to app_queue in the following order: * Hangup SIP/Agent * Hangup SIP/foo * Optimize L;1/L;2 * Hangup L;2 * Hangup L;1 When app_queue receives the hangup of the agent or the caller, it will attempt to publish the AgentComplete event. However, it now has a problem - it thinks its agent is the ;1 side of the Local channel, as it never received the optimization event. At the same time, that channel is already gone. This results in getting NULL from the Stasis cache. What's more, we can't really wait for the optimization message, as we are currently handling the hangup of the channel that the optimization event would tell us to use. This patch modifies the behavior in app_queue such that, since we still have a lot of pertinent queue information (interface, queue name, etc.), we now raise the event with what information we know. The channels involved now may or may not be present. Users will still at least get the "AgentComplete" event, which "completes" the known Agent information. Review: https://reviewboard.asterisk.org/r/2878/ (closes issue ASTERISK-22507) Reported by: Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a1c1fc3538	manager: Fix crash when appending a manager channel variable In r399887, a minor performance improvement was introduced by not allocating the manager variable struct if it wasn't used. Unfortunately, when directly accessing an ast_channel struct, manager assumed that the struct was always allocated. Since this was no longer the case, things got a bit crashy. This fixes that problem by simply bypassing appending variables if the manager channel variable struct isn't there. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	8a41aa9bf5	app_cdr and res_parking: Fix some resource leaks. * 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	10fcad8486	chan_sip: Increase some scratch buffer sizes dealing with caller id. * Eliminated an unnecessary initialization in check_user_full(). (closes issue ASTERISK-22477) Reported by: Michael Shepelev ........ Merged revisions 400013 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	1f908c58d0	res_pjsip: crash when using localnet and external_signaling_address options 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	ac5a46f7fa	chan_sip: Reject calls on 200 OKs if no SDP has been received When Asterisk receives a 200 OK in response to an invite, that peer should have sent an SDP at some point by then. If the channel has never received an SDP, media won't have been set and the remote address won't be known. Endpoints in general should not be doing this. This patch makes it so that Asterisk will simply hang up a call if it sends a 200 OK at this point. So far this odd behavior for endpoints has only been observed in tests which involved manually created SIP transactions in SIPp. (closes issue ASTERISK-22424) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2827/ ........ Merged revisions 399939 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399962 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	d865190a7e	astobj2: Remove OBJ_CONTINUE support. OBJ_CONTINUE was a strange feature that came into the world under suspicious circumstances to support an abuse of the ao2_container by chan_iax2. Since chan_iax2 no longer uses OBJ_CONTINUE, it is safe to remove it. The simplified code should help performance slightly and make understanding the code easier. Review: https://reviewboard.asterisk.org/r/2887/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	b2545b9ff9	Fix refleaks of ast_rtp_instance structures. These refleaks were causing bridged calls not to close their RTP ports. Thus a call would leave open 4 ports (RTP for party A, RTCP for party A, RTP for party B, and RTCP for party B). This led to an eventual depletion of available RTP ports. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	e087610307	Restore usefulness of the CEL Peer field This change makes the CEL peer field useful again for BRIDGE_ENTER and BRIDGE_EXIT events and fills the field with a comma-separated list of all channels in the bridge other than the channel that is entering or exiting the bridge. Review: https://reviewboard.asterisk.org/r/2840/ (closes issue ASTERISK-22393) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	a8b4e59dbc	pjsip: race condition in registrar 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	16d449ef38	Minor performance bump by not allocate manager variable struct if we don't need it git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	fa6f45f35a	Adding a few words to the Dial option 'r' help text to clarify its tone argument description git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f9ba38751b	chan_dahdi: CLI "core stop gracefully" has needless delay for PRI and SS7. The PRI and SS7 link control threads are not stopped correctly when the chan_dahdi.so module is unloaded. The link control threads pri_dchannel() and ss7_linkset() are not awakened from a poll() to cancel the thread. * Added a SIGURG signal after requesting the thread cancel to break the link control thread poll() immediately. For SS7 it was slightly worse, the link poll() timeout would always be whatever was the last libss7 scheduled event time used. If no libss7 scheduled event was pending, the thread could run more often than necessary. * Set nextms to 60 seconds for the ss7_linkset() poll() if there is no other libss7 scheduled event. ........ Merged revisions 399818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399834 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	921c0444e9	Broke the build - Fixing XML DTD violation added in r399782, missing <para> tags inside a <note> git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	b32560d657	chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok 1st Issue When a realtime peer sends an un-REGISTER request, Asterisk un-registers the peer but the database table record still has regseconds and fullcontact for the peer. This results in calls attempting to be routed to the peer which is no longer registered. The expected behavior is to get busy/congested when attempting to call an un-registered peer through the dialplan. What was discovered is that we are clearing out the peer's registration in the database in parse_register_contact() when calling expire_register() but then upon returning from parse_register_contact(), update_peer() is run which stores back in the database table regseconds and fullcontact. 2nd Issue The reporter pointed out that the 200 ok being returned by Asterisk after un-registering a peer contains a Contact header with ;expires= and the Expires header is not set to 0. This is actually a regression. Tests were created for this second issue (ASTERISK-22548). The tests have been reviewed and a Ship It! was received on those tests. This patch does the following: * Do not ignore the Expires header value even when it is set to 0. The patch sets the pvt->expiry earlier on in the function so that it is set properly and used. * If pvt->expiry is 0, do not call update_peer since that means the peer has already been un-registered and there is no need to update the database record again since nothing has changed. (closes issue ASTERISK-22428) Reported by: Ben Smithurst Tested by: Ben Smithurst, Michael L. Young Patches: asterisk-22428-rt-peer-update-and-expires-header.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2869/ ........ Merged revisions 399794 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399795 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	fb6b258197	Fixing documentation for the configOption "external_media_address" of both 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e134fb26d2	astobj2: Made use OBJ_SEARCH_xxx identifiers as field enum values internally. * Made ao2_unlink to protect itself from stray OBJ_SEARCH_xxx values passed in. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	5177f4a68f	chan_iax2: Prevent some needless breaking of the native IAX2 bridge. * Clean up some twisted code in the iax2_bridge() loop. * Add AST_CONTROL_VIDUPDATE and AST_CONTROL_SRCCHANGE to a list of frames to prevent the native bridge loop from breaking. * Passing the AST_CONTROL_T38_PARAMETERS frame should also allow FAX over a native IAX2 bridge. (issue ABE-2912) Review: https://reviewboard.asterisk.org/r/2870/ ........ Merged revisions 399697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399708 from http://svn.asterisk.org/svn/asterisk/branches/11 For v12 and above this is really just documentation until IAX2 native bridging is restored. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a99bc28c30	app_queue: Don't be quite so aggressive in initializing the array We only need the first character. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	576b9b982f	app_queue: Initialize array holding MixMonitor exec options If the channel variable MONITOR_EXEC is set, app_queue will pass the specified execution parameters to the MixMonitor application when a queue is recorded. If that channel variable is not set, the buffer that holds the escaped value was not being initialized to NULL, and so would be passed to the MixMonitor application with garbage. Hilarity ensued as app_mixmonitor attempted to execute gobeldy-gook. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c0576c7816	Fix a performance problem CDRs There is a large performance price currently in the CDR engine. We currently perform two ao2_callback calls on a container that has an entry for every channel in the system. This is done to create matching pairs between channels in a bridge. As such, the portion of the CDR logic that this patch deals with is how we make pairings when a channel enters a mixing bridge. In general, when a channel enters such a bridge, we need to do two things: (1) Figure out if anyone in the bridge can be this channel's Party B. (2) Make pairings with every other channel in the bridge that is not already our Party B. This is a two step process. In the first step, we look through everyone in the bridge and see if they can be our Party B (single_state_process_bridge_enter). If they can - yay! We mark our CDR as having gotten a Party B. If not, we keep searching. If we don't find one, we wait until someone joins who can be our Party B. Step 2 is where we changed the logic (handle_bridge_pairings and bridge_candidate_process). Previously, we would first find candidates - those channels in the bridge with us - from the active_cdrs_by_channel container. Because a channel could be a candidate if it was Party B to an item in the container, the code implemented multiple ao2_container callbacks to get all the candidates. We also had to store them in another container with some other meta information. This was rather complex and costly, particularly if you have 300 Local channels (600 channels!) going at once. Luckily, none of it is needed: when a channel enters a bridge (which is when we're figuring all this stuff out), the bridge snapshot tells us the unique IDs of everyone already in the bridge. All we need to do is: For all channels in the bridge: If the channel is us or our Party B that we got in step 1, skip it Compare us and the candidate to figure out who is Party A (based on some specific rules) If we are Party A: Make a new CDR for us, append it to our chain, and set the candidate as Party B If they are Party A: If they don't have a Party B: Make a new CDR for them, append us to their chain, and us as Party B Otherwise: Copy us over as Party B on their existing CDR. This patch does that. Because we now use channel unique IDs to find the candidates during bridging, active_cdrs_by_channel now looks up things using uniqueid instead of channel name. This makes the more complex code simpler; it does, however, have the drawback that dialplan applications and functions will be slightly slower as they have to iterate through the container looking for the CDR by name. That's a small price to pay however as the bridging code will be called a lot more often. This patch also does two other minor changes: (1) It reduces the container size of the channels in a bridge snapshot to 1. In order to be predictable for multi-party bridges, the order of the channels in the container must be stable; that is, it must always devolve to a linked list. (2) CDRs and the multi-party test was updated to show the relationship between two dialed channels. You still want to know if they talked - previously, dialed channels were always ignored, which is wrong when they have managed to get a Party B. (closes issue ASTERISK-22488) Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/2861/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	61c2cc894c	Fix crash in res_pjsip on load if error occurs, and prevent unloading of 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ec234be357	res_rtp_asterisk: Fix ref leaks in ast_rtcp_read(). Moved rtcp_report RAII_VAR declaration into the loop so it is unref'ed after every loop. Moved message_blob to loop and switched it to a regular variable. The regular variable was used since message_blob is used in a very contained way. (closes issue ASTERISK-22565) Reported by: Corey Farrell Patches: rtcp_report-leak.patch (license #5909) patch uploaded by Corey Farrell Tested by: Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1577e10111	media_index: Fix process_description_file() memory leak of file_id_persist. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	dd3acf4f57	features_config: Fix config ref leak of parkinglots. This leak happend for just about every channel created. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	5de3863884	app_queue: Fix json blob ref leak. The json ref from queue_member_blob_create() was never released. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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