asterisk

Commit Graph

Author	SHA1	Message	Date
Jenkins2	34b3665612	Merge "res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact" into 13	8 years ago
Rodrigo Ramírez Norambuena	cecf6540dc	cdr: fix mistake spelling of a word for Unanswered. Change-Id: I7a610bef369924523a445c7e849ee88cc45dc5df	8 years ago
Alexei Gradinari	8f356192d1	app_voicemail: IMAP connection control A new global option "imap_poll_logout" was added to specify whether need to disconnect from the IMAP server after polling of mailboxes. ASTERISK-27068 #close Closing IMAP connection after loading mailbox from voicemail.conf ASTERISK-24052 #close Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a	8 years ago
Jenkins2	507ce0aa95	Merge "res_stasis: Plug reference leak on stolen channels" into 13	8 years ago
George Joseph	470bccd769	Merge "app_voicemail: IMAP logout on reload/unload" into 13	8 years ago
Jenkins2	707e0e62e6	Merge "res_pjsip: New endpoint option "notify_early_inuse_ringing"" into 13	8 years ago
Alexei Gradinari	8e749c8f51	res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact If the endpoint's last contact is deleted unsolicited MWI has to be unsubscribed. ASTERISK-27051 #close Change-Id: I33e174e0b9dba0998927d16d6d100fda5c7254e0	8 years ago
Jenkins2	2c87fced4f	Merge "formats/format_g729: Fix typo in comment" into 13	8 years ago
Jenkins2	47b9651658	Merge "Core/PBX: Deadlock between dialplan execution and application unregistration." into 13	8 years ago
George Joseph	edfdb4dff5	res_stasis: Plug reference leak on stolen channels When a stasis channel is stolen by another app, the control structure is unreffed but never unlinked from the app_controls container. This causes the channel reference to leak. Added OBJ_UNLINK to the callback in channel_stolen_cb. Also added some additional channel lifecycle debug messages to channel.c. ASTERISK-27059 #close Repoorted-by: George Joseph Change-Id: Ib820936cd49453f20156971785e7f4f182c56e14	8 years ago
Matthew Fredrickson	0a40073750	formats/format_g729: Fix typo in comment There was a typo in a comment. This commit is to fix the typo. ASTERISK-27060 #close Change-Id: Ic2699f8dbeaacd58ccb6ec3203e853e1babe3235	8 years ago
Alexei Gradinari	a6e4899612	res_pjsip: New endpoint option "notify_early_inuse_ringing" This option was added to control whether to notify dialog-info state 'early' or 'confirmed' on Ringing when already INUSE. The value "yes" is useful for some SIP phones (Cisco SPA) to be able to indicate and pick up ringing devices. ASTERISK-26919 #close Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711	8 years ago
Jenkins2	0b7a0681a3	Merge "res_ari: Add "module loaded" check to ari stubs" into 13	8 years ago
Jenkins2	d88344c3d4	Merge "chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read." into 13	8 years ago
Jan Friesse	005a4afa6b	res_corosync: Change thread stack size In Corosync 2.x libraries were changed to use LibQB IPC. Sadly LibQB IPC doesn't support copy-free access to received buffer, so Corosync libraries were rewritten to use stack as buffer. Mostly the needed stack size is quite small, but for all *_dispatch functions, 1MiB is needed. Asterisk function ast_pthread_create_background set stack size for new thread to much smaller AST_BACKGROUND_STACKSIZE (~500KiB). This results in Asterisk crash when running with Corosync 2.x. Patch solves this issue by creating it's own version of ast_pthread_create_background which sets stack size to much higher value (actually it's AST_BACKGROUND_STACKSIZE + 3MiB). Another problem may appear when "corosync show members" netconsole command is executed. It is also executed in thread and also has only 500KiB stack size. Sadly it calls corosync_cfg_get_node_addrs which again needs at least 1MiB stack. Solution is to use HAVE_COROSYNC_CFG_STATE_TRACK as a discriminator between Corosync 1.x and 2.x. If 1.x is found, nothing changes. If 2.x is found, NodeID is displayed instead of IP address. ASTERISK-25370 #close Reported by: mdu113 Change-Id: Id95b0d21ab6e708e7d74ad8786c587211676fa08	8 years ago
George Joseph	7901b9853e	res_ari: Add "module loaded" check to ari stubs The recent change to make the use of LOAD_DECLINE more consistent caused res_ari to unload itself before declining if the ari.conf file wasn't found. The ari stubs though still tried to use the configuration resulting in segfaults. This patch creates a new CHECK_ARI_MODULE_LOADED macro which tests to see if res_ari is actually loaded and causes the stubs to also decline if it isn't. The macro was then added to the mustache template's "load_module" function. ASTERISK-27026 #close Reported-by: Ronald Raikes Change-Id: I263d56efa628ee3c411bdcd16d49af6260c6c91d	8 years ago
Jenkins2	37bc343b58	Merge "channel: Fix reference counting in ast_channel_suppress." into 13	8 years ago
Alexei Gradinari	3b6c327c51	app_voicemail: IMAP logout on reload/unload Closing IMAP connection on module reload or unload. ASTERISK-24052 #close Change-Id: I2a40182aa9ef249fa6865d33570430e9ada68525	8 years ago
Jenkins2	ae6c38db98	Merge "res_pjsip_pubsub: Fix reference to released endpoint" into 13	8 years ago
Jenkins2	5fa52f0b5a	Merge "bridge: Add a deferred queue." into 13	8 years ago
Richard Mudgett	b9a4ab8c8c	chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read. The construction of the returned string assumed incorrectly that the supplied buffer would always be initialized as an empty string. If it is not an empty string we could overrun the supplied buffer by the length of the non-empty buffer string plus one. It is also theoreticaly possible for the supplied buffer to be overrun by a string terminator during a read operation even if the supplied buffer is an empty string. * Fix the assumption that the supplied buffer would already be an empty string. The buffer is not guaranteed to contain an empty string by all possible callers. * Fix string terminator buffer overrun potential. Change-Id: If6a0806806527678c8554b1dcb34fd7808aa95c9	8 years ago
Jenkins2	8b12ad2f61	Merge "app_voicemail: IMAP logout on MWI unsubscribe" into 13	8 years ago
Jenkins2	a3b142dbe0	Merge "app_voicemail.c: Fix compile error when IMAP enabled." into 13	8 years ago
Jenkins2	320fb81580	Merge "res_pjsip_refer/session: Calls dropped during transfer" into 13	8 years ago
Joshua Colp	4910a3bf40	channel: Fix reference counting in ast_channel_suppress. The ast_channel_suppress function wrongly decremented the reference count of the underlying structure used to keep track of what should be suppressed on a channel if the function was called multiple times on the same channel. This change cleans up the reference counting a bit so this no longer occurs. ASTERISK-27016 Change-Id: I2eed4077cb4916e6626f9f120b63b963acc5c136	8 years ago
George Joseph	8a858275eb	Merge "res_rtp_asterisk: Fix ssrc change for rtcp srtp" into 13	8 years ago
Jenkins2	ffedf68136	Merge "res_pjsip_session: Correct inverted test in session_outgoing_nat_hook" into 13	8 years ago
Jenkins2	4ade9085d5	Merge "res_pjsip_transport_websocket: Add NULL check in get_write_timeout" into 13	8 years ago
George Joseph	3065bcd021	Merge "pjproject_bundled: Use the asterisk github mirror for download" into 13	8 years ago
Joshua Colp	4ece39f476	Merge "BuildSystem: Add patches to allow building with recent LibreSSL" into 13	8 years ago
Richard Mudgett	f1a209d5ac	app_voicemail.c: Fix compile error when IMAP enabled. Change-Id: I2703f15b4099b4210c68eccf293105d1975c1fc1	8 years ago
Frederic LE FOLL	dc307af7f2	Core/PBX: Deadlock between dialplan execution and application unregistration. Not easy to reproduce, but we have noticed deadlocks when unloading a module while dialplan is handling a request. The deadlock is between : 1) Dialplan execution: pbx_extension_helper() first taking conlock, then pbx_findapp() [when called] asking for lock on apps list. 2) Application unregistration: ast_unregister_application() first taking lock on apps list, then unreference_cached_app() [when called] asking for conlock. As a protection, I suggest to modify ast_unregister_application(), so that it anticipates the need of conlock, before taking the lock on apps list. The side effect is a longer unavailability of conlock when unregistering an application. ASTERISK-27041 Change-Id: I0db0f1eb320da6a5758cce3a47d765be1face8e2	8 years ago
George Joseph	c2eea791e4	res_pjsip_pubsub: Fix reference to released endpoint destroy_subscription was attempting to get the id of the subscription tree's endpoint after we'd already called ao2_cleanup on it causing a segfault. Moved the cleanup until after the debug statement and since endpoint could also be NULL at this point, check for that as well. ASTERISK-27057 #close Reported-by: Ryan Smith Change-Id: Ice0a7727f560cf204d870a774c6df71e159b1678	8 years ago
George Joseph	2dee95cc7a	res_pjsip_session: Correct inverted test in session_outgoing_nat_hook There was a typo introduced in commit `776ffd77` which was preventing the transport's external media address from being used. ASTERISK-27024 #close Reported-by: Christopher van de Sande patches: patch.diff submitted by Florian Floimair (license 6892) Change-Id: I7ec617171eaa2d86d2680b00cf37d5088adafc27	8 years ago
Jenkins2	e26d15fabb	Merge "CFLAGS for BIND8 support" into 13	8 years ago
Jørgen H	e16a669c70	res_pjsip_transport_websocket: Add NULL check in get_write_timeout Added check for NULL return value when calling ast_sorcery_retrieve_by_id in function get_write_timeout ASTERISK-27046 Change-Id: I9357717278da631c3a1cb502c412693929b0cb41	8 years ago
Jenkins2	44d4b55697	Merge "res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled" into 13	8 years ago
George Joseph	7dafe82751	res_rtp_asterisk: Fix ssrc change for rtcp srtp It looks like there was a copy/paste error in ast_rtp_change_source where if there was a rtcp srtp instance, instead of updating its ssrc we were updating the srtp instance ssrc twice. ASTERISK-27022 #close Reported-by: Michael Walton Change-Id: Ic88f3aee7227b401c58745ac265ff92c19620095	8 years ago
Joshua Colp	e414833f6e	bridge: Add a deferred queue. This change adds a deferred queue to bridging. If a bridge technology determines that a frame can not be written and should be deferred it can indicate back to bridging to do so. Bridging will then requeue any deferred frames upon a new channel joining the bridge. This change has been leveraged for T.38 request negotiate control frames. Without the deferred queue there is a race condition between the bridge receiving the T.38 request negotiate and the second channel joining and being in the bridge. If the channel is not yet in the bridge then the T.38 negotiation fails. A unit test has also been added that confirms that a T.38 request negotiate control frame is deferred when no other channel is in the bridge and that it is requeued when a new channel joins the bridge. ASTERISK-26923 Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415	8 years ago
Kevin Harwell	6cdf3191d3	res_pjsip_refer/session: Calls dropped during transfer When doing an attended transfer it's possible for the transferer, after receiving an accepted response from Asterisk, to send a BYE to Asterisk, which can then be processed before Asterisk has time to start and/or complete the transfer process. This of course causes the transfer to not complete successfully, thus dropping the call. This patch makes it so any BYEs received from the transferer, after the REFER, that initiate a session end are deferred until the transfer is complete. This allows the channel that would have otherwise been hung up by Asterisk to remain available throughout the transfer process. ASTERISK-27053 #close Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a	8 years ago
George Joseph	0bde568669	pjproject_bundled: Use the asterisk github mirror for download We now mirror the pjproject tarball and md5 at https://github.com/asterisk/third-party/tree/master/pjproject To improve download reliability, we now get the tarball from our mirror instead of from pjsip.org. ASTERISK-27052 #close Reported-by: 'alex' Change-Id: I60236587a8935bfa71fcc391f4e2ecb31918c08a	8 years ago
Joshua Colp	bc51d4324a	Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing." into 13	8 years ago
Alexei Gradinari	08be5e01e8	app_voicemail: IMAP logout on MWI unsubscribe Closing IMAP connection on MWI unsubscribe. ASTERISK-24052 #close Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd	8 years ago
Joshua Colp	d715beae16	Merge "codecs.conf.sample: Fix max_bandwidth speling error" into 13	8 years ago
Joshua Colp	052cd049bc	Merge "BuildSystem: Fix build on FreeBSD due to missing crypt.h" into 13	8 years ago
Jenkins2	1f7aedf6f7	Merge "eventfd: Disable during cross compilation" into 13	8 years ago
Alexei Gradinari	59c9bbe696	res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled If sending unsolicited mwi to all endpoints on startup is disabled (mwi_disable_initial_unsolicited=yes) do not need to create subscriptions. If there are many (thousands) realtime endpoints configured with unsolicited mwi and Vociemail Storage configured as ODBC or IMAP there will be huge number of DB/IMAP requests on startup. ASTERISK-26230 #close Change-Id: I50ae909639e3ee298b931a54def4b2b9e0fb86c5	8 years ago
David M. Lee	68de35a6a0	CFLAGS for BIND8 support Some systems (like macOS) require BIND_8_COMPAT to be defined so that the nameser libraries are, well, BIND8 compatible. Change-Id: If79fc27a64f90de1835b5aa3aadfa9be22bd16b0	8 years ago
Sean Bright	da3312457e	codecs.conf.sample: Fix max_bandwidth speling error Reported by Sylvain Boily via asterisk-dev mailing list. Change-Id: Idc7623f335aea3e144dd369ba383b9a757480a9d	8 years ago
Guido Falsi	6a64f65fe6	BuildSystem: Add patches to allow building with recent LibreSSL Add some #if defined checks which allow building against LibreSSL. These patchess come from OpenBSD ports: https://cvsweb.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/patches/ ASTERISK-27043 #close Reported by: OpenBSD ports Change-Id: I2f6c08a5840b85ad4d2b75370b947ddde7a9a572	8 years ago

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