asterisk

Commit Graph

Author	SHA1	Message	Date
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
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	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
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
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
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
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
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
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
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
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
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
Joshua Colp	7144c739e9	res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable. This change adds a configuration option which adds a 'user=phone' parameter if the user portion of the request URI or the From URI is determined to be a number. Review: https://reviewboard.asterisk.org/r/4073/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	86a4ce4957	PJSIP: Enforce module load dependencies This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub have loaded properly before attempting to load any modules that depend on them since the module loader system is not currently capable of resolving module dependencies on its own. ASTERISK-24312 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/4062/ ........ Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425691 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	bfee1b4bc5	res_rtp_asterisk: Fix a bug where ICE state would get reset when it shouldn't. In the case where the ICE negotiation had not yet started current state would get wiped when it shouldn't. This also removes channel binding as in practice this does not work well with other implementations. ........ Merged revisions 425644 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425645 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425646 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	3d58066de9	parking_tests: Fix assertions and possibly crashes in res_parking unit tests Assertions were caused by attempting to play music on hold to a channel with no formats. Parking unit test channels were given formats and a technology so that they would be able to pretend to read/write frames. ASTERISK-24413 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4075/ ........ Merged revisions 425611 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	527b58aeb7	res_phoneprov: Create accessor for ast_phoneprov_std_variable_lookup Based on feedback from Richard, I created an accessor for res_phoneprov/ast_phoneprov_std_variable_lookup and added load priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/ ........ Merged revisions 425480 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425481 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	fbb19db0c8	res_fax: Fix reference leak caused by gateway sessions Fax gateway session objects can be re-used, causing the same gateway session to be added to faxregistry.container more than once. This change causes fax_session_new to remove the reserved session from the container before it's id is changed, ensuring it's possible for the session to be freed. ASTERISK-24392 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4049/ ........ Merged revisions 425457 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425458 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425459 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	01bdc80475	res_fax: Resolve module reference leak caused by reserved sessions Remove reference to module providing reserved session after adding a reference to the final module. This re-reference is done to ensure that module references are correct even if the final session selects a different module than the reserved session. ASTERISK-18923 #close Reported by: Grigoriy Puzankin Review: https://reviewboard.asterisk.org/r/4048/ ........ Merged revisions 425405 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425407 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425411 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425415 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c7e6b6ba3d	manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	8d6f1d763c	res_rtp_asterisk: Make the ICE transport check case insensitive as some implementations use 'udp'. ........ Merged revisions 425360 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425361 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425362 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	2a03efdbae	res_calendar_ews: Relax neon version check to work with 0.30 too. Allow res_calendar_ews to work not only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close Reported by: Tzafrir Cohen Review: https://reviewboard.asterisk.org/r/4068/ ........ Merged revisions 425286 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425287 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425288 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425289 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	6a3c11c75b	res_phoneprov: Cleanup module load error handling Tested module load/reload interaction between res_phoneprov and res_pjsip_phoneprov_provider in cases where res_phoneprov didn't load correctly (usually misconfiguration or missing phoneprov.conf) Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4069/ ........ Merged revisions 425264 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425265 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c3ff212cae	res/res_phoneprov: Bail on registration if res_phoneprov didn't load If res_phoneprov failed to fully load (due to not being configured), the providers container will be NULL. If a module attempts to register a phone provisioning provider, it should check for the presence of the container. If there is no providers container, it should return an error. This patch makes the ast_phoneprov_provider_register function do that... otherwise this would be a silly commit message. ........ Merged revisions 425220 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425221 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	c46100ad5f	res_pjsip_phoneprov_provider: Add missing dependency on pjproject. ........ Merged revisions 425216 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425217 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425218 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0ef680cff0	res_pjsip_nat: Place source port into rport of responses if 'force_rport' is on. When the 'force_rport' option is enabled the behavior should be the same as if the remote side placed rport into the message themselves. Therefore any responses we send should include the source port of the request in the rport of the Via header. #SIPit31 ASTERISK-24387 #close Reported by: Matt Jordan ........ Merged revisions 425131 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425132 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	aef63118da	res_pjsip_phoneprov_provider: fix compile breakage on AST_VECTOR endpoint->inbound_auths was changed to a vector in 13 and I committed the 12 patch instead of the 13 patch. Tested-by: George Joseph ........ Merged revisions 425052 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	6fc4df7279	res_rtp_asterisk: Crash if no candidates received for component When starting ice if there is not at least one remote ice candidate with an RTP component asterisk will crash. This is due to an assertion in pjnath as it expects at least one candidate with an RTP component. Added a check to make sure at least one candidate contains an RTP component and at least one candidate has an RTCP component. ASTERISK-24383 #close Review: https://reviewboard.asterisk.org/r/4039/ ........ Merged revisions 425031 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c6837c236f	res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprov This module allows res_pjsip to integrate with res_phoneprov. It handles the pjsip 'phoneprov' object type. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3976/ ........ Merged revisions 425007 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425008 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	3a187aa14a	res/res_phoneprov: Don't cancel Asterisk load on module load failure ........ Merged revisions 424985 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424986 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	cc595f7353	res_phoneprov: Refactor phoneprov to allow pluggable config providers This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424964 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	8b0089ea1d	res_rtp_asterisk: Allow only UDP ICE candidates. The underlying library, pjnath, that res_rtp_asterisk uses for ICE support does not have support for ICE-TCP. As candidates are passed through directly to it this can cause error messages to occur when it receives something unexpected (such as a TCP candidate). This change merely ignores all non-UDP candidates so they never reach pjnath. ASTERISK-24326 #close Reported by: Joshua Colp ........ Merged revisions 424852 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424853 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424854 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a9011106b6	res/res_pjsip_outbound_registration: Initialize auth_reject_permanent parameter Prior to this patch, the auth_reject_permanent parameter was not initialized on the registration client state, leading to the parameter being disabled regardless of the value specified in pjsip.conf. This patch initialized the setting on the registration client state to the provided configuration value. ASTERISK-24398 #close ........ Merged revisions 424730 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424731 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	523da7d1b3	res/res_pjsip_pubsub: Fix typo in WARNING message ........ Merged revisions 424713 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cce3d99ec8	res_pjsip/pjsip_options: Do not 404 an OPTIONS request not sent to an endpoint An OPTIONS request that is sent to Asterisk but not to a specific endpoint is currently sent a 404 in response. This is because, not surprisingly, an empty extension is never going to be found in the dialplan. This patch makes it so that we only attempt to look up the endpoint in the dialplan if it is specified in the OPTIONS request URI. #SIPit31 ASTERISK-24370 #close Reported by: Matt Jordan ........ Merged revisions 424624 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424625 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	45b7b474ac	res_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible (although it shouldn't occur) for pjproject to pass up an rdata object with a NULL msg in the msg_info. Needless to say, things that attempt to dereference this are in for a rough ride. In particular, this caused crashes in three different locations, all of which are 'low level' enough to intercept an rdata object early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3) res_pjsip/distributor Anything that can intercept an rdata object before res_pjsip/distributor should be defensive when looking at the received packet. #SIPit31 ASTERISK-24369 #close Reported by: Matt Jordan ........ Merged revisions 424618 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424619 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f27f41a288	res/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions A subscription that has been persisted can - for various reasons - fail to be re-created on startup. This patch resolves a number of crashes that occurred when a subscription cannot be re-created on several off-nominal paths. #SIPit31 ASTERISK-24368 #close Reported by: Matt Jordan ........ Merged revisions 424601 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424602 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	24ded9d9eb	res_pjsip: Fix XML typo and update CHANGES. ASTERISK-24199 ........ Merged revisions 424528 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424529 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0165c5f95a	chan_pjsip: Fix deadlock when masquerading PJSIP channels. Performing a directed call pickup resulted in a deadlock when PJSIP channels were involved. A masquerade needs to hold onto the channel locks while it swaps channel information between the two channels involved in the masquerade. With PJSIP channels, the fixup routine needed to push a fixup task onto the PJSIP channel's serializer. Unfortunately, if the serializer was also processing a task that needed to lock the channel, you get deadlock. * Added a new control frame that is used to notify the channels that a masquerade is about to start and when it has completed. * Added the ability to query taskprocessors if the current thread is the taskprocessor thread. * Added the ability to suspend/unsuspend the PJSIP serializer thread so a masquerade could fixup the PJSIP channel without using the serializer. ASTERISK-24356 #close Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/4034/ ........ Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424472 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b1f8eba178	PJSIP: Restore functional default for callerid_privacy The pjsip config option default fixups from r424263 altered the functional default from "allowed_not_screened" to "allowed". This change restores the functional default value when none is provided. ........ Merged revisions 424426 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424427 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2b0777c017	res_pjsip: Make transport cipher option accept a comma separated list of cipher names. Improvements to the res_pjsip transport cipher option. * Made the cipher option accept a comma separated list of OpenSSL cipher names. Users of realtime will be glad if they have more than one name to list. * Added the CLI command 'pjsip list ciphers' so a user can know what OpenSSL names are available for the cipher option. * Updated the cipher option online XML documentation to specify what is expected for the value. * Updated pjsip.conf.sample to not indicate that ALL is acceptable since ALL does not imply a preference order for the ciphers and PJSIP does not simply pass the string to OpenSSL for interpretation. ASTERISK-24199 #close Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/4018/ ........ Merged revisions 424393 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424394 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	a752ca00bd	res_hep: Release allocation reference to configuration. ASTERISK-24362 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4026/ ........ Merged revisions 424312 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424313 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	adba2a8d7f	res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash. During the latest update to DTLS-SRTP support the ability to configure the hash used for fingerprints was added. This gave us two supported ones: SHA-1 and SHA-256. The default was accordingly updated to SHA-256. Unfortunately this configuration ability was not exposed within res_pjsip. This change adds a dtls_fingerprint option that controls it. #SIPit31 ........ Merged revisions 424290 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424291 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

1 2 3 4 5 ...

2724 Commits (97e1c7f3a9223981ee910bb820bb4c5fe70d18e9)