asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	bd42a09d7f	config: BUG: Restore ability for non-templ to be used as base objs in config. My recent refactor of config.c accidentally removed the capability for an object to inherit from a non-template object. This patch restores the capability to inherit from both template and non-template objects. Tested-by: George Joseph Reported-by: Scott Griepentrog ASTERISK-24487 #close Review: https://reviewboard.asterisk.org/r/4147/ ........ Merged revisions 427227 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427228 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	97e1c7f3a9	func_talkdetect: Fix stasis message leak in audiohook callback. ASTERISK-24482 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4142/ ........ Merged revisions 427203 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427204 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	9f2874639d	res_http_websockets: Fix extra unref of module In websocket_add_protocol_internal is used to add the "echo" protocol, but ast_websocket_remove_protocol is used to remove it. This causes an extra call to ast_module_unref. ASTERISK-24480 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4140/ ........ Merged revisions 427200 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427201 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	bdc35c77b9	Fix crash caused by merge error on review 4138 When merging from 12 to 13 there were conflicts, I mistakenly had the loop run ast_closestream(others[0]) when it should be ast_closestream(others[x]). ........ Merged revisions 427181 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d159885e50	res_pjsip_outbound_registration: Add virtual line support. Virtual line support establishes a relationship between messages related to an outbound registration and a local endpoint. This is accomplished by attaching a parameter to the Contact of the outbound registration and looking for it on any received requests. If the parameter exists and can be matched to an outbound registration the configured endpoint is associated with the request. Review: https://reviewboard.asterisk.org/r/2964/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	33f0251b6c	res_pjsip: Add disable_tcp_switch option. When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In some circumstances (on some networks), this can cause some issues with messages not getting sent to the correct destination - and can also cause connections to get dropped due to quirks in pjproject deciding to terminate TCP connections with no messages. While fixing the routing/messaging issues is important, having a configuration option in Asterisk that tells pjproject to not switch over to TCP would be useful. That way, if some glitch is discovered on some other network/site, we can at least disable the behavior until a fix is put into place. AFS-197 #close Review: https://reviewboard.asterisk.org/r/4137/ ........ Merged revisions 427129 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427130 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	b9aeff9580	chan_pjsip: Update CHANGES file to include 'moh_passthrough' setting git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ac091d4184	chan_pjsip: Add support for passing hold and unhold requests through. This change adds an option, moh_passthrough, that when enabled will pass hold and unhold requests through using a SIP re-invite. When placing on hold a re-invite with sendonly will be sent and when taking off hold a re-invite with sendrecv will be sent. This allows remote servers to handle the musiconhold instead of the local Asterisk instance being responsible. Review: https://reviewboard.asterisk.org/r/4103/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	285be15aaf	Fix compile error caused by review 4138 There is no procedure called ast_closeframe, fix code to use ast_closestream. Reported By: Matt Jordan ........ Merged revisions 427087 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427088 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427089 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	509c04ef38	Fix ast_writestream leaks Fix cleanup in __ast_play_and_record where others[x] may be leaked. This was caught where prepend != NULL && outmsg != NULL, once realfile[x] == NULL any further others[x] would be leaked. A cleanup block was also added for prepend != NULL && outmsg == NULL. 11+: Fix leak of ast_writestream recording_fs in app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/ ........ Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427024 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427025 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427026 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	85c1822a9d	func_jitterbuffer: fix frame leaks. Fix code paths where it is possible for frames to leak. Fix uninitialized variable in jb_get_fixed and jb_get_adaptive. ASTERISK-22409 #related Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4128/ ........ Merged revisions 427019 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427020 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427021 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5db1c978e3	res/res_stasis: Fix crash on module unload while performing operation When the res_stasis module is unloaded, it will dispose of the apps_registry container. This is a problem if an ARI operation is in flight that attempts to use the registry, as the shutdown occurs in a separate thread. This patch adds some sanity checks to the various routines that access the registry which cause the operations to fail if the apps_registry does not exist. Crash caught by the Asterisk Test Suite. ........ Merged revisions 426995 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426996 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	4219c40775	install init.d files on GNU/kFreeBSD Review: https://reviewboard.asterisk.org/r/4118/ ........ Merged revisions 426926 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426927 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426933 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426934 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	28173ddf05	pjsip: clarify tls cert and key file usage A question arose as to whether a .pem file could be provided in place of the .crt and .key files in a PJSIP TLS configuration. I tested this and discovered that although a cert will be read from the pem file, a key will not, and thus the priv_key_file entry is still required. This update to the fine documentation clarifies the option usage. AST-1448 #close Review: https://reviewboard.asterisk.org/r/4129/ Reported by: John Bigelow ........ Merged revisions 426928 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426930 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	f59db388a7	pjsip: Handle outbound unregister correctly This updates the status of the outbound registration to reflect when it has been unregistered. Since the registration is unregistered but is not stopped, the registration schedule remains active as before. The patch also updates the documentation of both the AMI and CLI commands. ASTERISK-24411 #close Review: https://reviewboard.asterisk.org/r/4119/ Reported by: John Bigelow patches: unregister-patch1.txt uploaded by John Bigelow (License 5091) ........ Merged revisions 426923 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426924 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d88282af40	channels/sip/reqresp_parser: Fix unit tests for r426594 When r426594 was made, it did not take into account a unit test that verified that the function properly populated the unsupported buffer. The function would previously memset the buffer if it detected it had any contents; since this function can now be called iteratively on successive headers, the unit tests would now fail. This patch updates the unit tests to reset the buffer themselves between successive calls, and updates the documentation of the function to note that this is now required. ........ Merged revisions 426858 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426860 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426863 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426865 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	bf684b63a3	REF_DEBUG: Install refcounter.py to $(ASTDATADIR)/scripts This change ensures refcounter.py is installed to a place where it can be found by the Asterisk testsuite if REF_DEBUG is enabled. ASTERISK-24432 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4094/ ........ Merged revisions 426830 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426831 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426832 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426833 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	e4374a3abe	app_queue: fix a couple leaks to struct call_queue in set_member_value set_member_value has a couple leaks to references in the variable q found through testsuite tests/queues/set_penalty. Also remove the REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible with the updated REF_DEBUG code. ASTERISK-24466 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4125/ ........ Merged revisions 426805 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426806 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426807 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ced81afff2	audiohooks: Clean references to formats Cleanup references to in_translate[x].format and out_translate[x].format in ast_audiohook_detach_list. ASTERISK-24465 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4124/ ........ Merged revisions 426803 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	a537e314d1	res_pjsip_exten_state: PJSIPShowSubscriptionsInbound causes crash Currently, it is possible for some subscriptions to get into a NULL state. When this occurs and the PJSIPShowSubscriptionsInbound ami action is issued and a device is subscribed for extension state then the associated subscription state object can't be located. The code then attempts to dereference a NULL object. Added a NULL check to avoid the problem. Reported by: John Bigelow ........ Merged revisions 426779 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426780 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	cd52456ea1	res_pjsip: incorrect qualify statistics after disabling for contact When removing the qualify_frequency from an AoR or a contact the statistics shown when issuing "pjsip show aors" from the CLI are incorrect. This patch deletes the contact's status object from sorcery, disassociating it from the contact, if the qualify_freqency is removed from configuration. ASTERISK-24462 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4116/ ........ Merged revisions 426755 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426757 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	5d8d90c402	app_voicemail: Fix unchecked bounds of myArray in IMAP_STORAGE. In update_messages_by_imapuser(), messages were appended to a finite array which resulted in a crash when an IMAP mailbox contained more than 256 entries. This memory is now dynamically increased as needed. Observe that this patch adds a bunch of XXX's to questionable code. See the review (url below) for more information. ASTERISK-24190 #close Reported by: Nick Adams Tested by: Nick Adams Review: https://reviewboard.asterisk.org/r/4126/ ........ Merged revisions 426691 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426692 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426696 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426702 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	c866ced76b	Add additional checks for NULL pointers to fix several crashes reported.	11 years ago
Matthew Jordan	0ddc3bde24	channels/chan_sip: Add improved support for 4xx error codes This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER response handling. This helps interoperability in a number of scenarios. Review: https://reviewboard.asterisk.org/r/3437 patches: rb3437.patch uploaded by oej (License 5267) ........ Merged revisions 426599 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426600 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426601 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426602 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ff83ff564c	channels/chan_sip: Support mutltiple Supported and Required headers A SIP request may contain multiple Supported: and Required: headers. Currently, chan_sip only parses the first Supported/Required header it finds. This patch adds support for multiple Supported/Required headers for INVITE requests. Review: https://reviewboard.asterisk.org/r/2478 ASTERISK-21721 #close Reported by: Olle Johansson patches: rb2478.patch uploaded by oej (License 5267) ........ Merged revisions 426594 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426595 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426596 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426597 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	8a69aedd17	Fix building chan_phone on big endian systems A left over from the formats conversion (Corey Farrell). ASTERISK-24458 #close Review: https://reviewboard.asterisk.org/r/4117/ ........ Merged revisions 426570 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0ed8aebda9	bridge_builtin_features: Add missing channel locks around ast_get_chan_features_general_config(). The feature_automonitor() and feature_automixmonitor() functions were not locking the channel around ast_get_chan_features_general_config(). Accessing the channel datastore list without the channel locked is a good way to corrupt the list or follow the pointer chain into oblivion. ........ Merged revisions 426531 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426552 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	7205d76d7d	res_fax: Resolve T38 gateway frame leak. When frames are translated by a fax gateway they need to be freed. The existing call to ast_frfree was unreachable. This change reorganizes fax_gateway_framehook to ensure that ast_frfree is called when needed. ASTERISK-24457 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4115/ ........ Merged revisions 426527 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426528 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426529 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	67e496c275	manager: Unsubscribe from acl_change_sub at shutdown. ASTERISK-24453 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4110/ ........ Merged revisions 426524 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426525 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Malcolm Davenport	1fe22c411d	ASTERISK-23512, correct inaccurate comment in manager.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8e9f593e3a	main/bridge: Destroy features struct on off nominal path during bridge impart When a channel is imparted to a bridge, the invocation of the function may provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart, the caller must assume that ownership has passed to the function, as in all paths the function destroys the struct prior to returning (as its purpose is to configure the behavior of the channel while in the bridge). On one off nominal path - where the channel already has a PBX thread - the struct was not being destroyed. This patch fixes that glitch. ASTERISK-24437 #close Reported by: Scott Griepentrog ........ Merged revisions 426431 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426432 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f4b4d42630	main/manager: Fix typo in AMI event documentation of "OriginateResponse" The parameter name is "Response", not "Resonse". ASTERISK-24430 #close Reported by: Dafi Ni ........ Merged revisions 426366 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426367 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426368 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Malcolm Davenport	68d9872f58	ASTERISK-24323, fix bug in documentation of AGI STREAM FILE CONTROL git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Malcolm Davenport	684b8762a9	ASTERISK-24419, fix incorrect syntax for setting language in extensions.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	2290393273	app_queue: Cleanup ao2_iterator Clean ao2_iterator, resolving reference leak to queue members. ASTERISK-24454 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4111/ ........ Merged revisions 426255 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426260 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426266 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ab16f46139	func_cdr: Fix CDR_PROP payload leak Remove duplicate allocation of payload, preventing leak. ASTERISK-24455 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4113/ ........ Merged revisions 426252 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	ef8cdd40e5	configure: Add autoconf check for libopus. Because opus transcoding support cannot be included in the standard Asterisk distribution, a few codec_opus implementations have popped up. To make it easier for people to drop in opus support in their own installations, this patch adds configure checks for libopus. Review: https://reviewboard.asterisk.org/r/4106/ ........ Merged revisions 426234 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5a17878085	res/res_http_websocket: Fix minor nits found by wdoekes on r409681 When Moises committed the fixes for WSS (which was a great patch), wdoekes had a few style nits that were on the review that got missed. This patch resolves what I think were all of the ones that were still on the review. Thanks to both moy for the patch, and wdoekes for the reviews. Review: https://reviewboard.asterisk.org/r/3248/ ........ Merged revisions 426209 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426210 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426211 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	62bee9b327	res/res_phoneprov: Fix crash on shutdown caused by container cleanup In res_phoneprov, unloading the module first destroys the http_routes container, followed by the users. However, users may have a route in the http_routes container; the validity of this container is not checked in the users destructor. Hence, we hit an assert as the container has already been set to NULL. This patch does two things: (1) It adds a sanity check in the user destructor (because why not) (2) It switches the order of destruction, so that users are disposed of prior to the HTTP routes they may hold a reference to. Note that this crash was caught by the Test Suite (go go testing!) ........ Merged revisions 426174 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426176 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	130a3fcd7f	res/res_srtp: Fix include issue for libsrtp 1.5.0 In libsrtp 1.5.0, crypto_get_random is no longer resolved simply by including srtp.h. Now, one must include crypto_kernel.h as well. As it turns out, this header file has been provided by the library since 2006, so this is a relatively benign change. ASTERISK-24436 #close Reported by: Patrick Laimbock ........ Merged revisions 426140 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426141 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426142 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426143 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c084728690	Documentation: Improve documentation for ExtensionStatus AMI events Review: https://reviewboard.asterisk.org/r/4085/ ........ Merged revisions 426120 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Shaun Ruffell	c4d7e7e270	codec_dahdi: Cannot use struct ast_translator.core_{src,src}_codec. This fixes a Segmentation fault introduced in r419044 "media formats: re-architect handling of media for performance improvements". The problem is that codec_dahdi was using core_src_codec and core_dst_codec in the ast_translator structure when these fields were never set. Now instead of trying to map the new core codec descriptions to the way DAHDI defines different codecs, we will store the DAHDI specific formats in 'struct translator' directly so we can refer to them without mapping. This also allows us to remove the "global_format_map" structure, since we can now query the list of translators directly to make sure we do not ever register a DAHDI based translator for a specific path more than once and eliminate the need to keep the list and the map in sync. ASTERISK-24435 #close Reported by: Marian Koniuszko Review: https://reviewboard.asterisk.org/r/4105/ ........ Merged revisions 426097 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2165868be7	translage.c: Fix regression when generating translation path strings. Fix the AMI Status action read and write translation path strings from growing for each channel in the status event list by reseting the ast string given to ast_translate_path_to_str() to fill in the given translation path. ........ Merged revisions 426079 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	dad0334cf1	AST-2014-011: Fix POODLE security issues There are two aspects to the vulnerability: (1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module to use TLSv1+. At this time, it does not refactor res_jabber/res_xmpp to use the TCP/TLS core, which should be done as an improvement at a latter date. (2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left unspecified, will default to the OpenSSL SSLv23_method. This method allows for all ecnryption methods, including SSLv2/SSLv3. A MITM can exploit this by forcing a fallback to SSLv3, which leaves the server vulnerable to POODLE. This patch adds WARNINGS if a user uses SSLv2/SSLv3 in their configuration, and explicitly disables SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly chosen. For TLS servers, Asterisk will no longer support SSLv2 or SSLv3. Much thanks to abelbeck for reporting the vulnerability and providing a patch for the res_jabber/res_xmpp modules. Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425 #close Reported by: abelbeck Tested by: abelbeck, opsmonitor, gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded by mjordan (License 6283) ........ Merged revisions 425987 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425991 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	5e10e369b1	build: Force -fsigned-char on platforms where the default for char is unsigned gcc on the ARM platform defaults 'char' to 'unsigned char' whereas Intel and SPARC default to 'signed char'. This is only an issue in the rare cases where negative values are assigned to a 'char' but this this patch insures compatibility by detecting platforms that default to 'unsigned' and adding an '-fsigned-char' flag to _ASTCFLAGS. If compiling for ARM (native or cross-compile) be sure to run ./bootstrap.sh and ./configure to regenerate the build files. You shouldn't have to do this for Intel or SPARC. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4091/ ........ Merged revisions 425964 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425965 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425966 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	404b6ab3ab	res/res_pjsip_sdp_rtp: Revert 425924 This patch for r425924 introduced a bug, wherein sending an INVITE request with no SDP would cause Asterisk to not send an SDP Offer in the 200 OK. The current structure of res_pjsip_sdp_rtp is a bit hard to deal with to fix this, as create_outgoing_sdp has no knowledge of whether or not it is creating an SDP as a new Offer or an Answer. This is something of an oversight in the callback definition, as the caller of it does have this information. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b263c8bdae	res/res_pjsip_sdp_rtp: Remove left over reference to override_prefs The usage of the local override_prefs variable in create_outgoing_sdp_stream was previously to track an override format preference set by PJSIP_MEDIA_OFFER. Now, however, that function simply sets the joint capabilities structure, session->req_caps. During the media format rework, the override_prefs was instead used to check if there were any formats in session->req_caps. However, this usage isn't useful in create_outgoing_sdp_stream. session->req_caps contains the negotiated formats for all streams, not just the current one being created. Thus, so long as any stream of any type has provided a format, override_prefs will be non-zero. Hence, its usage in checking whether or not we should look at the formats on the endpoint or the joint capabilities is generally useless. There's only two things useful to check: (1) Does the endpoint have a format for the media type? (2) Did we negotiate a format for the media type? If either of those is a 'no', then we must kill the media stream. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	b8f687f27c	Sample Configurations: make 'pjsip reload' reload all reloadable pjsip modules AST-1432 #close Reported by: John Bigelow ........ Merged revisions 425905 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8f58592252	res_pjsip_session/res_pjsip_sdp_rtp: Be more tolerant of offers When an inbound SDP offer is received, Asterisk currently makes a few incorrection assumptions: (1) If the offer contains more than a single audio/video stream, Asterisk will reject the entire stream with a 488. This is an overly strict response; generally, Asterisk should accept the media streams that it can accept and decline the others. (2) If the offer contains a declined media stream, Asterisk will attempt to process it anyway. This can result in attempting to match format capabilities on a declined media stream, leading to a 488. Asterisk should simply ignore declined media streams. (3) Asterisk will currently attempt to handle offers with AVPF with use_avpf=No/AVP with use_avpf=Yes. This mismatch results in invalid SDP answers being sent in response. If there is a mismatch between the media type being offered and the configuration, Asterisk must reject the offer with a 488. This patch does the following: * Asterisk will accept SDP offers with at least one media stream that it can use. Some WARNING messages have been dropped to NOTICEs as a result. * Asterisk will not accept an offer with a media type that doesn't match its configuration. * Asterisk will ignore declined media streams properly. #SIPit31 Review: https://reviewboard.asterisk.org/r/4063/ ASTERISK-24122 #close Reported by: James Van Vleet ASTERISK-24381 #close Reported by: Matt Jordan ........ Merged revisions 425868 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425879 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0d0e38a0e1	res_pjsip_keepalive: Add runtime configurable keepalive module for connection-oriented transports. This change adds a module which is configurable using the keep_alive_interval setting in the global section that will send a CRLF keep alive to all active connection-oriented transports at the provided interval. This is useful because it can help keep connections open through NATs. This functionality also exists within PJSIP but can not be controlled at runtime and requires recompiling it. Review: https://reviewboard.asterisk.org/r/4084/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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