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Author	SHA1	Message	Date
Joshua Colp	c8f3074cc4	res_sorcery_config: Improve object lookup times. ... The res_sorcery_config module currently uses a fixed bucket size of 53. This means that depending on the number of objects you either end up with excess buckets or a lot of collisions. Due to the way that res_sorcery_config is implemented it's actually possible to make the bucket size dynamic based on the number of objects. This is due to the fact that each loading of the config file produces a new container and does not modify the existing one. This change uses the number of expected objects and finds a prime number near it. In practice depending on the number of objects this can speed up lookups anywhere from 2X to 15X. This change also removes the lock from the container as it is not needed. Review: https://reviewboard.asterisk.org/r/4423/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	a3044cbf02	res_pjsip: Add "pjsip show version" CLI command. ... When debugging things it can be useful to know absolutely what version of pjproject res_pjsip is running against. This change adds a "pjsip show version" CLI command which can be used to query for this. ASTERISK-24685 #close Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/4424/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	ce70587ba6	res_timing_pthread: Fix leaky pipes. ... During some refactoring the way private information for timers was stored was changed. As a result of this the action which normally removed the timer upon closure in res_timing_pthread was also removed causing the timer to remain after it should using up resources. This change ensures that the timer is removed upon closure. ASTERISK-24768 #close Reported by: Matthias Urlichs patches: timer.patch submitted by Matthias Urlichs (license 5508) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	4f4d03fdd1	apps/app_mixmonitor: Move Test Event for MIXMONITOR_END to after it finishes ... The Test Event for MIXMONITOR_END - which signals that a MixMonitor has completed - technically fired before the filestream was closed. If a test used this to trigger a condition to verify that the file was written, it could result in a race condition where the file size would not be what the test expected. Luckily, no tests were using this (although they should have been). Since the test event needed to be moved after the point where the MixMonitor autochan has been destroyed, the test event no longer emits the channel name. Luckily, nothing needs it. ........ Merged revisions 431788 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	758a897876	sorcery: Output an error message if a wizard is specified for an object type and it isn't found. ... ASTERISK-24612 #close Reported by: Joshua Colp git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	8c6e3ad3b4	res_pjsip_exten_state: Improve log message when a subscription is attempted to a non-existent extension. ... ASTERISK-24716 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431754 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	3543a36362	'information' ends with an 'n'. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	5d26236758	chan_pjsip: Fix crash when CHANNEL dialplan function is invoked with pjsip argument and no type. ... ASTERISK-24771 #close Reported by: Niklas Larsson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	4d797f17c5	res_pjsip_session: Fix double re-INVITE collision crash. ... A multi-asterisk box setup with direct media enabled would occasionally crash when two re-INVITE collisions on a call leg happen in a row. The re-INVITE logic only had one timer struct to defer the re-INVITE. When the second collision happens the timer struct is overwritten and put into the timer heap again. Resources for the first timer are leaked and the heap has two positions occupied by the same timer struct. Now the heap ordering is potentially corrupted, the timer will fire twice, and any resources allocated for the second timer will be released twice. * The solution is to put the collided re-INVITE into the delayed requests queue with all the other delayed requests and cherry pick the next request that can come off the queue when an event happens. * Changed to put delayed BYE requests at the head of the delayed queue. There is no sense in processing delayed UPDATEs and re-INVITEs when a BYE has been requested. * Made the start of a BYE request flush the delayed requests queue to prevent a delayed request from overlapping the BYE transaction. I saw a few cases where a delayed re-INVITE got started after the BYE transaction started. * Changed the delayed_request struct to use an enum instead of a string for the request method. Cherry picking the queue is easier with an enum than string comparisons and the compiler can warn if a switch statement does not cover all defined enum values. * Improved the debug output to give more information. It helps to know which channel is involved with an endpoint. Trunks can have many channels associated with the endpoint at the same time. ASTERISK-24727 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	1995baad71	ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app ... This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	e8ec15a9ef	res_pjsip: dtls_handler causes Asterisk to crash ... There have been a couple of times where a crash occurred in the dtls_handler section of the code for res_pjsip. Unfortunately, in working this issue the problem was unable to be reproduced. After looking at the backtraces and through the code the current best guess as to why this happened might be due to a reentrance problem and the strtok function. So, the current fix is to convert the strtok function into the reentrant version of the function, strtok_r. ASTERISK-24741 #close Reported by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4409/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	e64d151fae	ari_websockets: removed extra check on websocket session read ... When merging the websocket timeout issue (ASTERISK-24701) an extra, almost duplicate, check was left in the code that should not have been. This removes it. ASTERISK-24701 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4412/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	feddab7944	HTTP: Stop accepting requests on final system shutdown. ... There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	29f3ff0b61	channels/chan_sip: Fix RealTime error during SIP unregistration with MariaDB ... When a SIP device that has its registration stored in RealTime unregisters, the entry for that device is updated with blank values, i.e., "", indicating that it is no longer registered. Unfortunately, one of those values that is 'blanked' is the device's port. If the column type for the port is not a string datatype (the recommended type is integer), an ODBC or database error will be thrown. MariaDB does not coerce empty strings to a valid integer value. This patch updates the query run from chan_sip such that it replaces the port value with a value of '0', as opposed to a blank value. This is the value that other database backends coerce the empty string ("") to already, and the handling of reading a RealTime registration value from a backend already anticipates receiving a port of '0' from the backends. ASTERISK-24772 #close Reported by: Richard Miller patches: chan_sip.diff uploaded by Richard Miller (License 5685) ........ Merged revisions 431673 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	72e5ba2ce8	res_http_websocket: websocket write timeout fails to fully disconnect ... When writing to a websocket if a timeout occurred the underlying socket did not get closed/disconnected. This patch makes sure the websocket gets disconnected on a write timeout. Also a notice is logged stating that the websocket was disconnected. ASTERISK-24701 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4412/ ........ Merged revisions 431669 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	2531f75057	Enable REF_DEBUG for ast_module_ref / ast_module_unref. ... Add ast_module_shutdown_ref for use by modules that can only be unloaded during graceful shutdown. When REF_DEBUG is enabled: * Add an empty ao2 object to struct ast_module. * Allocate ao2 object when the module is loaded. * Perform an ao2_ref in each place where mod->usecount is manipulated. * ao2_cleanup on module unload. ASTERISK-24479 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4141/ ........ Merged revisions 431662 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
George Joseph	4d8ab20a8a	res_pjsip_config_wizard: Add ability to auto-create hints. ... Looking at the Super Awesome Company sample reminded me that creating hints is just plain gruntwork. So you can now have the pjsip conifg wizard auto-create them for you. Specifying 'hint_exten' in the wizard will create 'exten => <hint_exten>,hint/PJSIP/<wizard_id>' in whatever is specified for 'hint_context'. Specifying 'hint_application' in the wizard will create 'exten => <hint_exten>,1,<hint_application>' in whatever is specified for 'hint_context'. The default for 'hint_context' is the endpoint's context. There's no default for 'hint_application'. If not specified, no app is added. There's no default for 'hint_exten'. If not specified, neither the hint itself nor the application will be created. Some may think this is the slippery slope to users.conf but hints are a basic necessity for phones unlike voicemail, manager, etc that users.conf creates. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4383/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	32e42e50cc	res/ari/resource_channels: Add missing 'no_answer' reason to DELETE /channels ... One of the canonical reasons for hanging up a channel is because the far end failed to answer - or because someone else answered, and we want to get rid of this channel. This patch adds the missing value to the 'reason' query parameter for the DELETE /channels operation. Review: https://reviewboard.asterisk.org/r/4400 ASTERISK-24745 #close Reported by: Ben Merrills patches: add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills (License 6678) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	03445a147e	Blocked revisions 431620 ... While it may not be obvious, r431620 should not occur in Asterisk 13. * We no longer set the SIP_DEFER_BYE_ON_TRANSFER flag during the handling of the INVITE with Replaces. This is now set and handled explicitly in the attended transfer and blind transfer code. * An INVITE with Replaces replacing a channel in a Bridge will now safely eject the channel being replaced. No masquerade occurs. * An INVITE with Replaces replacing a channel not in a Bridge will masquerade, but will do so in such a fashion that we can ensure that we are hanging up the channel when completed. Since the code the patch fixes no longer exists due to core framework changes, we should send a BYE naturally without the need for the flag. ........ channels/chan_sip: Ensure that a BYE is sent during INVITE w/Replaces transfer Consider a scenario where Alice and Bob have an established dialog with each other external to Asterisk. Bob decides to perform an attended transfer of Alice to Asterisk. In this case, Alice will send an INVITE with Replaces to Asterisk, where the Replaces specifies Bob's dialog with Asterisk. In this particular scenario, Asterisk will complete the transfer, but - since Bob's channel has had Alice masqueraded into it and is now a Zombie - a BYE request will not be sent. This patch fixes that issue by adding a new flag to chan_sip that tracks whether or not we have an INVITE with Replaces. If we do, the flag is used on the sip_pvt to ensure that a BYE request is sent, even if the channel has been masqueraded away. Review: https://reviewboard.asterisk.org/r/4362/ ASTERISK-22436 #close Reported by: Eelco Brolman Tested by: Jeremiah Gowdy, Kristian Høgh patches: asterisk-11-hangup-replaced-3.diff uploaded by Jeremiah Gowdy (License 6358) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431621 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	8582411344	res/res_odbc: Remove unneeded queries when determining if a table exists ... This patch modifies the ast_odbc_find_table function such that it only performs a lookup of the requested table if the table is not already known. Prior to this patch, a queries would be executed against the database even if the table was already known and cached. Review: https://reviewboard.asterisk.org/r/4405/ ASTERISK-24742 #close Reported by: ibercom patches: patch.diff uploaded by ibercom (License 6599) ........ Merged revisions 431617 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	675b2b8103	res/res_pjsip_sdp_rtp: Fix leak of local ICE candidates when applying to SDP ... When an SDP is created for an outgoing request/response, the ICE candidates obtained from the RTP instance are currently leaked. This causes the ao2 container that holds the candidates to never properly be reclaimed when the RTP instance is destroyed. This patch properly decrements the ICE candidates' container if it is successfully obtained. ASTERISK-24769 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	323c0927ac	various: cleanup issues found during leak hunt ... In this collection of small patches to prevent Valgrind errors are: fixes for reference leaks in config hooks, evaluating a parameter beyond bounds, and accessing a structure after a lock where it could have been already free'd. Review: https://reviewboard.asterisk.org/r/4407/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	18c8c1bae3	res_pjsip_keepalive: Don't crash if PJSIP module is not loaded. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	2f2eb1931a	sorcery: Don't try to load object types which haven't been defined. ... The act of defining wizards for an object type in sorcery.conf will create a minimal object type. This can cause a problem when a module has multiple sorcery instances (which all get the wizards from sorcery.conf applied) but the sorcery instances do not all contain full information about the object types. Upon loading errors will occur stating that the objects can not be created. This is confusing and is actually perfectly fine. This change makes it so that only object types which have been fully defined will be loaded. ASTERISK-24748 #close Reported by: Joshua Colp git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	f67402a52a	res_format_attr_h264: Fix crash when determining joint capability. ... The res_format_attr_h264 module currently incorrectly attempts to copy SPS and PPS information from the wrong attribute. This change fixes that. ASTERISK-24616 #close Reported by: Yura Kocyuba Review: https://reviewboard.asterisk.org/r/4392/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	05e2832b35	app_agent_pool: Fix initial module load agent device state reporting. ... When the app_agent_pool module initially loads there is a race condition between the thread loading agents.conf and the device state internal processing thread. If the device state internal processing thread handles the agent creation state updates before the thread that loaded agents.conf registers the device state provider callback then the cached agent state is "Invalid". When a consumer module like app_queue asks for the agent state it gets the cached "Invalid" state instead of the real state from the provider. * Moved loading the agents.conf configuration to the last thing setup by app_agent_pool in load_module(). Now the device state provider callback is registered before the config is loaded so the agent creation state updates are guaranteed to get the initial device state. * Removed some now redundant config cleanup on error in load_config(). * Added lock protection when accessing the device state in agent_pvt_devstate_get() and eliminated the RAII_VAR() usage. ASTERISK-24737 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4390/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	6583b4de98	res_pjsip_outbound_publish: eventually crashes when no response is ever received ... When Asterisk attempts to send SIP outbound publish information and no response is ever received (no 200 okay, 412, 423) the system eventually crashes. A response is never received because the system Asterisk is attempting to send publish information to is not available. The underlying pjsip framework attempts to send publish information. After several attempts it calls back into the Asterisk outbound publish code. At this point if the "client->queue" is empty Asterisk attempts to schedule a refresh which utilizes "rdata" and since no response was received the given "rdata" struture is NULL. Attempting to dereference a NULL object of course results in a crash. The fix here removes the dependency on rdata for schedule_publish_refresh. Instead param->expiration is now passed to it as this is set to -1 if no response is received. Also added a notification when no response is received. ASTERISK-24635 #close Reported by: Marco Paland Review: https://reviewboard.asterisk.org/r/4384/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Ashley Sanders	112d23c73e	HTTP: For httpd server, need option to define server name for security purposes ... Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. ASTERISK-24316 #close Reported By: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4374/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	43dd42d8ae	Fix some memory leaks. ... These memory leaks were found and fixed by John Hardin. I'm just committing them for him. ASTERISK-24736 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4389 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	f7d23dfcc6	stasis transfer: fix stasis bridge push race part two ... When swapping a Local channel in place of one already in a bridge (to complete a bridge attended transfer), the channel that was swapped out can actually be hung up before the stasis bridge push callback executes on the independant transfer thread. This results in the stasis app loop dropping out and removing the control that has the the app name which the local replacement channel needs so it can re-enter stasis. To avoid this race condition a new push_peek callback has been added, and called from the ast_bridge_impart thread before it launches the independant thread that will complete the transfer. Now the stasis push_peek callback can copy the stasis app name before the swap channel can hang up. ASTERISK-24649 Review: https://reviewboard.asterisk.org/r/4382/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	e8896ac008	Use SIPS URIs in Contact headers when appropriate. ... RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific scenarios when we are required to use SIPS URIs in Contact headers. Asterisk's non-compliance with this could actually cause calls to get dropped when communicating with clients that are strict about checking the Contact header. Both of the SIP stacks in Asterisk suffered from this issue. This changeset corrects the behavior in res_pjsip/chan_pjsip.c Review: https://reviewboard.asterisk.org/r/4345 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	22fc3359da	Use SIPS URIs in Contact headers when appropriate. ... RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific scenarios when we are required to use SIPS URIs in Contact headers. Asterisk's non-compliance with this could actually cause calls to get dropped when communicating with clients that are strict about checking the Contact header. Both of the SIP stacks in Asterisk suffered from this issue. This changeset corrects the behavior in chan_sip. ASTERISK-24646 #close Reported by Stephan Eisvogel Review: https://reviewboard.asterisk.org/r/4346 ........ Merged revisions 431423 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	b8ea23b0d1	Allow disabling of 100rel support on PJSIP endpoints. ... Due to an inversion error, setting 100rel=no would not actually change the current value of the setting (which defaulted to "yes"). With this fix, the inversion is corrected. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431420 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	6e5eb9af88	res_pjsip_exten_state: Reduce log clutter... change a WARNING to a VERBOSE/2 ... Reduce log clutter by changing the "Watcher for hint %s (removed|deactivated)" message from WARNING to VERBOSE/2. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4387/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e0461290d0	res_rtp_asterisk: Fix DTLS when used with OpenSSL 1.0.1k ... A recent security fix for OpenSSL broke DTLS negotiation for many applications. This was caused by read ahead not being enabled when it should be. While a commit has gone into OpenSSL to force read ahead on for DTLS it may take some time for a release to be made and the change to be present in distributions (if at all). As enabling read ahead is a simple one line change this commit does that and fixes the issue. ASTERISK-24711 #close Reported by: Jared Biel ........ Merged revisions 431384 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	8c068fc096	Fix file descriptor leak in RTP code. ... SIP requests that offered codecs incompatible with configured values could result in the allocation of RTP and RTCP ports that would not get reclaimed later. ASTERISK-24666 #close Reported by Y Ateya Review: https://reviewboard.asterisk.org/r/4323 AST-2015-001 ........ Merged revisions 431300 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	25a67d561c	Multiple revisions 431297-431298 ... ........ r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan 2015) | 17 lines Mitigate possible HTTP injection attacks using CURL() function in Asterisk. CVE-2014-8150 disclosed a vulnerability in libcURL where HTTP request injection can be performed given properly-crafted URLs. Since Asterisk makes use of libcURL, and it is possible that users of Asterisk may get cURL URLs from user input or remote sources, we have made a patch to Asterisk to prevent such HTTP injection attacks from originating from Asterisk. ASTERISK-24676 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4364 AST-2015-002 ........ r431298 | mmichelson | 2015-01-28 11:12:49 -0600 (Wed, 28 Jan 2015) | 3 lines Fix compilation error from previous patch. ........ Merged revisions 431297-431298 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431299 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	c3add776af	media formats: update res_format_attr_opus & silk ... In r419044, we changed how formats were handled, but the return value of the format_parse_sdp_fmtp functions in res_format_attr_opus and res_format_attr_silk were not updated, causing calls to fail. Ran into this when getting codec_opus working with Asterisk 13. Once the return value was corrected, we were crashing in opus_getjoint because of NULL format attributes. I've fixed this as well in this patch. Review: https://reviewboard.asterisk.org/r/4371/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	88fbe4e917	res_pjsip_outbound_registration: Fix reload race condition. ... Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	61822e78ae	tcptls: Bad file descriptor error when reloading chan_sip ... While running through some scenarios using chan_sip and tcp a problem would occur that resulted in a flood of bad file descriptor messages on the cli: tcptls.c:712 ast_tcptls_server_root: Accept failed: Bad file descriptor The message is received because the underlying socket has been closed, so is valid. This is probably happening because unloading of chan_sip is not atomic. That however is outside the scope of this patch. This patch simply stops the logging of multiple occurrences of that message. ASTERISK-24728 #close Reported by: Thomas Thompson Review: https://reviewboard.asterisk.org/r/4380/ ........ Merged revisions 431218 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	e2b493b8f0	chan_sip: stale nonce causes failure ... When refreshing (with a small expiration) a registration that was sent to chan_sip the nonce would be considered stale and reject the registration. What was happening was that the initial registration's "dialog" still existed in the dialogs container and upon refresh the dialog match algorithm would choose that as the "dialog" instead of the newly created one. This occurred because the algorithm did not check to see if the from tag matched if authentication info was available after the 401. So, it ended up assuming the original "dialog" was a match and stopped the search. The old "dialog" of course had an old nonce, thus the stale nonce message. This fix attempts to leave the original functionality alone except in the case of a REGISTER. If a REGISTER is received if searches for an existing "dialog" matching only on the callid. If the expires value is low enough it will reuse dialog that is there, otherwise it will create a new one. ASTERISK-24715 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4367/ ........ Merged revisions 431187 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	9e3d316dd1	res_pjsip: make it unloadable (take 2) ... Due to the original patch causing memory corruptions it was removed until the problem could be resolved. This patch is the original patch plus some added locking around stasis router subcription that was needed to avoid the memory corruption. Description of the original problem and patch (still applicable): The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4363/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	eda125f98d	app_confbridge: Repeatedly starting and stopping recording ref leaks the recording channel. ... Starting and stopping conference recording more than once causes the recording channels to be leaked. For v13 the channels also show up in the CLI "core show channels" output. * Reworked and simplified the recording channel code to use ast_bridge_impart() instead of managing the recording thread in the ConfBridge code. The recording channel's ref handling easily falls into place and other off nominal code paths get handled better as a result. ASTERISK-24719 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4368/ Review: https://reviewboard.asterisk.org/r/4369/ ........ Merged revisions 431135 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	b64f4bb6ee	bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media. ... This change fixes two issues: 1. During a swap operation bridging added the new channel before having the swap channel leave. This was not handled in bridge_native_rtp and could result in a channel not getting reinvited back to Asterisk. After this change the swap channel will leave first and the new channel will then join. 2. If a re-invite was received after a session had been established any upstream elements (such as bridge_native_rtp) were not notified that they may want to re-evaluate things. After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs and upstream can react. AST-1524 #close Review: https://reviewboard.asterisk.org/r/4378/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	a620b287bd	Manager: Fix Manager Action ModuleLoad to give correct response when reloading ... Prior to this patch, ModuleLoad would respond with an error indicating that the requested module wasn't found in spite of finding and reloading the module. Review: https://reviewboard.asterisk.org/r/4373/ ASTERISK-24721 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	7f9b28b0c6	ARI: Improve wiki documentation ... This patch improves the documentation of ARI on the wiki. Specifically, it addresses the following: * Allowed values and allowed ranges weren't documented. This was particularly frustrating, as Asterisk would reject query parameters with disallowed values - but we didn't tell anyone what the allowed values were. * The /play/id operation on /channels and /bridges failed to document all of the added media resource types. * Documentation for creating a channel into a Stasis application failed to note when it occurred, and that creating a channel into Stasis conflicts with creating a channel into the dialplan. * Some other minor tweaks in the mustache templates, including italicizing the parameter type, putting the default value on its own sub-bullet, and some other nicities. Review: https://reviewboard.asterisk.org/r/4351 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1a17693789	app_confbridge: Restore user's menu name to CLI output of 'confbridge list' ... When issuing a 'confbridge list XXXX' CLI command, the resulting output no longer displays the menu associated with a ConfBridge participant. The issue was caused by ASTERISK-22760. When that patch was done, it removed the copying of the menu name associated with the user from the actual user profile. This patch fixes the issue by copying the menu name over to the user profile when the menu hooks are applied to the user. Since that function now does a little bit more than just apply the hooks, the name of the function has been changed to cover the copying of the menu name over as well. In addition, there is a disparity between the menu name length as it is stored on the conf_menu structure and the confbridge_user structure; this patch makes the lengths match so that a strcpy can be used. Review: https://reviewboard.asterisk.org/r/4372/ ASTERISK-24723 #close Reported by: Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ceedd40370	res_parking: Fix crash due to race condition when unloading. ... There is currently a race condition when unloading the res_parking module. Depending on the will of the universe the subscription invocation may occur AFTER the module is unloaded. This is because the module does NOT use stasis_unsubscribe_and_join when terminating the subscription. It merely uses stasis_unsubscribe. This change makes it use stasis_unsubscribe_and_join which is documented for usage in this exact scenario. AST-1520 #close Review: https://reviewboard.asterisk.org/r/4375/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
David M. Lee	702d79de2a	Various fixes for OS X ... This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Fixed compilation issues with res_timing_kqueue (although tests still fail on OS X). * Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by: George Joseph ASTERISK-24544 #close Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/4327/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1fc823c770	dynamic realtime: Updates fail to work due to update fields being passed over ... When a crash was fixed due to usage of the REALTIME function in r423003, a regression was introduced into ast_update2_realtime where the update fields passed to the function would be skipped and the lookup field processed twice. The use of this function is a bit interesting: A variable argument list is used with two sentinel values - the first marks the end of the lookup fields/values; the second marks the end of the update fields/values. Unfortunately, ast_update2_realtime parses over the lookup fields twice, as opposed to parsing over the update fields. This causes the lookups to succeed, but the updates itself to have no effect. Note that the most common instance of this problem occurred in app_voicemail during the updating of a mailbox password. Thanks to the issue reporter, Paddy Grice, for pointing out the problem. Review: https://reviewboard.asterisk.org/r/4356/ ASTERISK-24231 ASTERISK-24626 #close Reported by: Paddy Grice git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago