asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	feb151979c	Merge "app_queue: Ensure member is removed from pending when hanging up." into certified/13.8	9 years ago
Joshua Colp	01e1ada455	app_queue: Ensure member is removed from pending when hanging up. When dialing channels it is possible that they may not ever leave the not in use state (Local channels in particular) by the time we cancel them. If this occurs but we know they were dialed we explicitly remove them from the pending members container so that subsequent call attempts occur. ASTERISK-26299 #close Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65 (cherry picked from commit `f69f5cd3c4`)	9 years ago
Joshua Colp	bbb041a1cf	app_queue: Only remove queue member from pending when state changes. It is possible for a not in use state change to occur multiple times causing a queue member to be removed from the pending call container prematurely. The first not in use state change will remove the queue member from the container. At this moment the member may be called and placed in the pending container. After this another not in use state change can be received which will remove it from the container. Despite being called at this point the code will incorrectly see that there are no pending calls to it. This change only removes it from the pending container if the state has actually changed. ASTERISK-26133 #close patches: app_queue.diff submitted by Richard Miller (license 5685) Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0 (cherry picked from commit `8a45461465`)	9 years ago
George Joseph	4574a24429	Merge "file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type" into certified/13.8	9 years ago
George Joseph	8a0ff36154	Merge "stasis_recording/stored: remove calls to deprecated readdir_r function." into certified/13.8	9 years ago
George Joseph	0d07070791	file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type One of the code paths in __ast_file_read_dirs will only get executed if the OS doesn't support dirent->d_type OR if the filesystem the particular file is on doesn't support it. So, while standard Linux systems support the field, some filesystems like XFS do not. In this case, we need to call stat() to determine whether the directory entry is a file or directory so we append the filename to the supplied directory path and call stat. We forgot to truncate path back to just the directory afterwards though so we were passing a complete file name to the callback in the dir_name parameter instead of just the directory name. The logic has been re-written to only create a full_path if we need to call stat() or if we need to descend into another directory. Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba	9 years ago
Kevin Harwell	7fc8b73f5c	stasis_recording/stored: remove calls to deprecated readdir_r function. The readdir_r function has been deprecated and should no longer be used. This patch removes the readdir_r dependency (replaced it with readdir) and also moves the directory search code to a more centralized spot (file.c) Also removed a strict dependency on the dirent structure's d_type field as it is not portable. The code now checks to see if the value is available. If so, it tries to use it, but defaults back to using the stats function if necessary. Lastly, for most implementations of readdir it should be thread-safe to make concurrent calls to it as long as different directory streams are specified. glibc falls into this category. However, since it is possible that there exist some implementations that are not safe, locking has been added for those other than glibc. ASTERISK-26412 ASTERISK-26509 #close Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba	9 years ago
Martin Tomec	7fb73f4963	app_queue: Ensure member is removed from pending when hanging up. In some cases member is added to pending_members, and the channel is hung up before any extension state change. So the member would stay in pending_members forever. So when we call do_hang, we should also remove member from pending. ASTERISK-26621 #close Change-Id: Iae476b5c06481db18ebe0fa594b3e80fdc9a7d54 (cherry picked from commit `d13be4eff6`)	9 years ago
Kevin Harwell	f0955f190a	Update for certified/13.8-cert4	9 years ago
Kevin Harwell	2e0239c28c	Update for certified/13.8-cert4	9 years ago
Walter Doekes	4fece22836	chan_sip: Do not allow non-SP/HTAB between header key and colon. RFC says SIP headers look like: HCOLON = ( SP / HTAB ) ":" SWS SWS = [LWS] ; sep whitespace LWS = [WSP CRLF] 1WSP ; linear whitespace WSP = SP / HTAB ; from rfc2234 chan_sip implemented this: HCOLON = ( LOWCTL / SP ) ":" SWS LOWCTL = %x00-1F ; CTL without DEL This discrepancy meant that SIP proxies in front of Asterisk with chan_sip could pass on unknown headers with \x00-\x1F in them, which would be treated by Asterisk as a different (known) header. For example, the "To\x01:" header would gladly be forwarded by some proxies as irrelevant, but chan_sip would treat it as the relevant "To:" header. Those relying on a SIP proxy to scrub certain headers could mistakenly get unexpected and unvalidated data fed to Asterisk. This change fixes so chan_sip only considers SP/HTAB as valid tokens before the colon, making it agree on the headers with other speakers of SIP. ASTERISK-26433 #close AST-2016-009 Change-Id: I78086fbc524ac733b8f7f78cb423c91075fd489b	9 years ago
Alexei Gradinari	016d20ce12	pjsip: Fix deadlock with suspend taskprocessor on masquerade If both channels which should be masqueraded are in the same serializer: 1st channel will be locked waiting condition 'complete' 2nd channel will be locked waiting condition 'suspended' On heavy load system a chance that both channels will be in the same serializer 'pjsip/distibutor' is very high. To reproduce compile res_pjsip/pjsip_distributor.c with DISTRIBUTOR_POOL_SIZE=1 Steps to reproduce: 1. Party A calls Party B (bridged call 'AB') 2. Party B places Party A on hold 3. Party B calls Voicemail app (non-bridged call 'BV') 4. Party B attended transfers Party A to voicemail using REFER. 5. When asterisk masquerades calls 'AB' and 'BV', a deadlock is happened. This patch adds a suspension indicator to the taskprocessor. When a session suspends/unsuspends the serializer it sets the indicator to the appropriate state. The session checks the suspension indicator before suspend the serializer. ASTERISK-26145 #close Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b	9 years ago
Mark Michelson	87e1ebc91a	res_pjsip_session: Do not call session supplements when it's too late. res_pjsip_sesssion was hooking into transaction and invite state changes. One of the reasons for doing so was due to the PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the message sending process, and so we should call session supplements to alter the outgoing message. In reality, this event was meant to indicate that the message either a) had already been sent, or b) required a DNS lookup and would be sent when the DNS query completed. In case (a), this meant we were altering an already-sent request/response for no reason. In case (b), this potentially meant we could be trying to alter a request/response at the same time that the DNS resolution completed. In this case, it meant we might be stomping on memory being used by the thread actually sending the message. This caused potential crashes and memory corruption. This patch removes the calls to session supplements from the case where the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to alter the message at this point is too late, and it can cause nothing but harm to try to do it. Because there were no longer any calls to the handle_outgoing() function, it has been removed. Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92 (cherry picked from commit `e043d1a55c`)	9 years ago
Joshua Colp	c8df727502	app_dial: Fix incorrect device state when channel is picked up. Given the scenario where multiple channels are dialed using Dial() but the caller is picked up using PickupChan() all outgoing channels except the channel specified to PickupChan() would be marked as ringing until the call had been hung up. When using the PickupChan application the channel executing the application is swapped into place of another channel. As part of this process the channel is answered. The Dial application has explicit logic which checks if the channel is answered, cancels all other outgoing channels, and bridges. This logic is different than the normal logic that is executed when an outgoing channel is answered. This different logic failed to publish dial events stating that the other outgoing channels had been canceled. As a result references to the outgoing channels were held onto by the dial masquerade process until the call had been ended and the channels had gone away. This would result in the channels appearing in the "core show channels" list despite not being present anymore and would also result in incorrect device state. This change makes it so that this logic also publishes dial events stating that the other outgoing channels have been canceled. ASTERISK-26549 Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f	9 years ago
Joshua Colp	4c50689d09	Revert "pjproject_bundled: Add patch to address SSL crash" This reverts commit `28cc8a9dff`. Change-Id: I777cf8173f7a88273090bed72bfe57fb0e72b84f	9 years ago
George Joseph	28cc8a9dff	pjproject_bundled: Add patch to address SSL crash Addresses crashes when an attempt is made to operate on an SSL socket after the socket has been closed. ASTERISK-26477 #close Change-Id: I421305b357558b4f9e690210dc0f4831ef4b3002 (cherry picked from commit `546ec4b038`)	9 years ago
Richard Mudgett	7c2bd702fd	Audit ast_json_pack() calls for needed UTF-8 checks. Added needed UTF-8 checks before constructing json objects in various files for strings obtained outside the system. In this case string values from a channel driver's peer and not from the user setting channel variables. * aoc.c: Fixed type mismatch in s_to_json() for time and granularity json object construction. ASTERISK-26466 Reported by: Richard Mudgett Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096	9 years ago
Richard Mudgett	3b1b2fc1d5	json: Check party id name, number, subaddresses for UTF-8. * Updated unit test as ast_json_name_number() is now NULL tolerant. ASTERISK-26466 #close Reported by: Richard Mudgett Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6	9 years ago
Richard Mudgett	a835adfa92	json: Add UTF-8 check call. Since the json library does not make the check function public we recreate/copy the function in our interface module. ASTERISK-26466 Reported by: Richard Mudgett Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99	9 years ago
Alexei Gradinari	7baedd9ecd	res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice. This patch removed call of pjsip_tx_data_dec_ref in send_notify if send_request failed. The pjsip_dlg_send_request deletes the message on error by itself. It seems this patch fixes next issues: ASTERISK-26199 ASTERISK-26166 ASTERISK-26174 Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a	9 years ago
George Joseph	a8e37c3d06	pjproject_bundled: Add MALLOC_DEBUG capability pjproject_bundled will now use the asterisk memory debugging APIs if MALLOC_DEBUG is turned on in menuselect. Because this required stubs for the executable programs and the python bindings, some Makefile reorganization was needed to properly handle the dependencies. As a result, the makefile now individually makes each of the pjproject libraries separately instead of making them all in 1 shot. The only visible change is that there are separate status lines printed for each library instead oif 1 for all libs. Also, the making of the pjproject dependency files was eliminated. They're not needed for building unless you're actively modifying pjproject source files and it makes the build process faster. Finally, any issues with parallel builds should be resolved again making the build faster. NOTE: The certified/13.8 version of this patch also builds libresample which is needed by pjsua. Later versions do not need libresample. Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0	9 years ago
Richard Mudgett	adcdecd47f	res_pjsip: Add ignore_uri_user_options option. This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62	9 years ago
George Joseph	ad7e072a6d	ChangeLog: Updated for certified/13.8-cert3	9 years ago
George Joseph	63f94b9d42	Release summaries: Add summaries for certified/13.8-cert3	9 years ago
George Joseph	6cec20200b	Release summaries: Remove previous versions	9 years ago
George Joseph	9c0f0eef6e	.version: Update for certified/13.8-cert3	9 years ago
George Joseph	3923b114b9	.lastclean: Update for certified/13.8-cert3	9 years ago
George Joseph	83362b5590	realtime: Add database scripts for certified/13.8-cert3	9 years ago
Joshua Colp	d947baa255	chan_sip: Don't allocate new RTP instances on top of old ones. In some scenarios dialog_initialize_rtp can be called multiple times on the same dialog. This can cause RTP instances to be leaked along with multiple file descriptors for each instance. This change makes it so the existing RTP instances are destroyed and not overwritten, stopping the memory leak. ASTERISK-26272 #close patches: ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73	9 years ago
Mark Michelson	df9aa402a5	ConfBridge: Make some announcements asynchronous. Confbridge announcements tend to block a channel while they are being played. In some circumstances, this is warranted since you want that particular channel not to hear the announcement (Example: "John Doe has entered the conference"). For others it makes less sense. This change first introduces methods for playing sounds asynchronously into the conference. This is very similar to how synchronous sounds are played, except the channel initiating the playback does not wait for the sound to complete before moving on. Asynchronous announcements are used for two circumstances: * Sounds played for a user after they have left the bridge * Sounds that play first to a single user and then the rest of the conference (if the channel and conference use the same language) ASTERISK-26289 #close Reported by Mark Michelson Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a	9 years ago
George Joseph	b1e827d259	res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_options ast_multicast_rtp_create_options now checks for NULL or empty options Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362	10 years ago
Mark Michelson	c218e038d7	ConfBridge: Rework announcer channel methodology NOTE: This patch was submitted earlier and reverted because of a failing test. The test has been patched so that it adjusts for the changes here, so this is being resubmitted for review. One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0	10 years ago
Joshua Colp	87969a6b0b	Merge "Revert "ConfBridge: Rework announcer channel methodology"" into certified/13.8	10 years ago
Joshua Colp	b82f0b7722	Revert "ConfBridge: Rework announcer channel methodology" This reverts commit `4ca730127c`. Change-Id: I8886feb69ae2dbf521a8c0937792349b70db52b2	10 years ago
Joshua Colp	66d50eccae	Merge "ConfBridge: Rework announcer channel methodology" into certified/13.8	10 years ago
Mark Michelson	4ca730127c	ConfBridge: Rework announcer channel methodology One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5	10 years ago
George Joseph	f40c6874c6	res_pjsip: Add contact_user to endpoint contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4	10 years ago
Kevin Harwell	78a6da0885	alembic/sqlalchemy: auto increment only allowed on a single column The extensions table defined two columns (id and priority) as primary key autoincrement columns. However only one is allowed when defining the primary key. This patch removes the autoincrement attribute from the priority column since it does not need to be as such and really should not have been on there in the first place. This patch also removes 'context', 'exten', and 'priority' from the primary key index and creates a new combined unique contraint index on them. ASTERISK-26183 #close Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b (cherry picked from commit `f6ec94cca6`)	10 years ago
Joshua Colp	ac0f73694b	ChangeLog: Updated for certified/13.8-cert2	10 years ago
Joshua Colp	2b4d367011	Release summaries: Add summaries for certified/13.8-cert2	10 years ago
Joshua Colp	d7eec92332	Release summaries: Remove previous versions	10 years ago
Joshua Colp	847b0d007d	.version: Update for certified/13.8-cert2	10 years ago
Joshua Colp	bd8581cd52	.lastclean: Update for certified/13.8-cert2	10 years ago
Joshua Colp	628620c5ef	realtime: Add database scripts for certified/13.8-cert2	10 years ago
Mark Michelson	13b950f4be	res_rtp_asterisk: Cache local RTCP address. When an RTCP packet is sent or received, res_rtp_asterisk generates a Stasis event that contains the RTCP report as well as the local and remote addresses that the report pertains to. The addresses are determined using ast_find_ourip(). For the local address, this will typically result in a lookup of the hostname of the server, and then a DNS lookup of that hostname. If you do not have the host in /etc/hosts, then this results in a full DNS lookup, which can potentially block for some time. This is especially problematic when performing RTCP reads, since those are done on the same thread responsible for reading and writing media. This patch addresses the issue by performing a lookup of the local address when RTCP is allocated. We then use this cached local address for the Stasis events when necessary. ASTERISK-26280 #close Reported by Mark Michelson Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556	10 years ago
Joshua Colp	acf021cdec	ChangeLog: Updated for certified/13.8-cert2-rc1	10 years ago
Joshua Colp	fba5cf4a00	Release summaries: Add summaries for certified/13.8-cert2-rc1	10 years ago
Joshua Colp	b2cc9b4879	Release summaries: Remove previous versions	10 years ago
Joshua Colp	20e25657fa	.version: Update for certified/13.8-cert2-rc1	10 years ago
Joshua Colp	08c26fba06	.lastclean: Update for certified/13.8-cert2-rc1	10 years ago

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27537 Commits (feb151979c53bc718e867701de7191eed45182cf) All Branches Search

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