asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin Harwell	a389f2d7a0	AST-2014-017 - app_confbridge: permission escalation/ class authorization. Confbridge dialplan function permission escalation via AMI and inappropriate class authorization on the ConfbridgeStartRecord action. The CONFBRIDGE dialplan function when executed from an external protocol (for instance AMI), could result in a privilege escalation. Also, the AMI action “ConfbridgeStartRecord” could also be used to execute arbitrary system commands without first checking for system access. The AMI “ConfbridgeStopRecord” has also been updated to only run under a system authorization. Asterisk now inhibits the CONFBRIDGE function from being executed from an external interface if the live_dangerously option is set to no. Also, the “ConfbridgeStartRecord” AMI action is now only allowed to execute under a user with system level access. ASTERISK-24490 Reported by: Gareth Palmer ........ Merged revisions 428332 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428334 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428339 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	1c88ca9d31	AST-2014-016: Fix crash when receiving an in-dialog INVITE with Replaces in res_pjsip_refer. The implementation of INVITE with Replaces in res_pjsip_refer did not expect them to occur in-dialog. As a result it would incorrectly attempt to hang up a channel it thought was under its control. In reality the channel would be under the control of another thread. When the other thread accessed the channel it would be accessing freed memory and could crash. This change makes res_pjsip_refer not act on an in-dialog INVITE with Replaces. ASTERISK-24528 #close Reported by: Joshua Colp ........ Merged revisions 428304 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428305 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d25eda5fb2	AST-2014-015: Fix race condition in chan_pjsip when sending responses after a CANCEL has been received. Due to the serialized architecture of chan_pjsip there exists a race condition where a CANCEL may be received and processed before responses (such as 180 Ringing, 183 Session Progress, and 200 OK) are sent. Since the session is in an unexpected state PJSIP will assert when this is attempted. This change makes it so that these responses are not sent on disconnected sessions. ASTERISK-24471 #close Reported by: yaron nahum ........ Merged revisions 428301 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428302 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	57c6f89bf0	stringfields: Fix bug in ast_string_fields_copy. ast_string_fields_copy relies on the fact that __ast_string_field_release_active never previously zeroed pool->used, so keeping the existing pointer was "ok". Now that existing pools can be reset to 'empty', it is important to set each field to __ast_string_field_empty after releasing the memory. ASTERISK-24535 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4186/ ........ Merged revisions 428272 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428273 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a7c9f4c668	ast_str: Fix improper member access to struct ast_str members. Accessing members of struct ast_str outside of the string manipulation API routines is invalid since struct ast_str is supposed to be treated as opaque. Review: https://reviewboard.asterisk.org/r/4194/ ........ Merged revisions 428244 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428245 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428246 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	7f8b7ace72	res_pjsip_sdp_rtp: Add support for optimistic SRTP. Optimistic SRTP is the ability to enable SRTP but not have it be a fatal requirement. If SRTP can be used it will be, if not it won't be. This gives you a better chance of using it without having your sessions fail when it can't be. Encrypt all the things! Review: https://reviewboard.asterisk.org/r/3992/ ........ Merged revisions 428222 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	b2e766a6b7	alembic: Fix alembic migration for 'moh_passthrough' option in res_pjsip. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	3119c3737f	res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI. There is no guarantee that when we get a Refer-To that it will be NULL terminated. As the URI parsing function requires it to be we now NULL terminate it. Additionally parsing the Refer-To as a 'To' header is needless and it can simply be done as a URI. This also fixes a problem where certain Refer-To headers would not be parsed as a 'To' header causing the REFER to fail. ASTERISK-24508 #close Reported by: Beppo Mazzucato Review: https://reviewboard.asterisk.org/r/4187/ ........ Merged revisions 428195 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428196 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a94efa239c	parking_tests.c: Add missing newline on a unit test message. ........ Merged revisions 428168 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428169 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428170 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2e750db120	Allow for transferer to retry when dialing an invalid extension. This allows for a configurable number of attempts for a transferer to dial an extension to transfer the call to. For Asterisk 13, the default values are such that upgrading between versions will not cause a behaivour change. For trunk, though, the defaults will be changed to be more user-friendly. Review: https://reviewboard.asterisk.org/r/4167 ........ Merged revisions 428145 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	4cea5fd4ba	chan_sip: Fix theoretical leak of p->refer. If transmit_refer is called when p->refer is already allocated, it leaks the previous allocation. Updated code to always free previous allocation during a new allocation. Also instead of checking if we have a previous allocation, always create a clean record. ASTERISK-15242 #close Reported by: David Woolley Review: https://reviewboard.asterisk.org/r/4160/ ........ Merged revisions 428117 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428118 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428119 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	948af7fd79	apps/app_confbridge: Ensure 'normal' users hear message when last marked leaves When r428077 was made for ASTERISK-24522, it failed to take into account users who are neither wait_marked nor end_marked. These users are also supposed to hear the 'leader has left the conference' message. Granted, this behaviour is a bit odd; however, that is how it used to work... and behaviour changes are not good. This patch ensures that if there are any 'normal' users present when the last marked user leaves the conference, the message will still be played to them. Note that this regression was caught by the Asterisk Test Suite's confbridge_nominal test, which has a quirky combination of users. ........ Merged revisions 428113 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428114 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428115 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fc2279afea	app_confbridge: Don't play leader leaving prompt if no one will hear it Consider the following: - A marked user in a conference - One or more end_marked only users in the conference When the marked users leaves, we will be in the conf_state_multi_marked state. This currently will traverse the users, kicking out any who have the end_marked flags. When they are kicked, a full ast_bridge_remove is immediately called on the channels. At this time, we also unilaterally set the need_prompt flag. When the need_prompt flag is set, we then playback a sound to the bridge informing everyone that the leader has left; however, no one is left in the bridge. This causes some odd behaviour for the end_marked users - they are stuck waiting for the bridge to be unlocked. This results in them waiting for 5 or 6 seconds of dead air before hearing that they've been kicked. Unfortunately, we do have to keep the bridge locked while we're playing back the 'leader-has-left' prompt. If there are any wait_marked users in the conference, this behaviour can't be easily changed - but we do make the case of the end_marked users better with this patch. Review: https://reviewboard.asterisk.org/r/4184/ ASTERISK-24522 #close Reported by: Matt Jordan ........ Merged revisions 428077 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428078 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428079 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	656601d8c4	chan_pjsip: Remove AOR check when dialing and one is specified. The AOR value may contain the name of an AOR or a full SIP URI. Checking if the AOR exists can't be done as a result of this. ........ Merged revisions 428051 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428052 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	bc02cbabd9	chan_sip: Fix bug where DTLS configuration from general would copy dtlsenable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6993743b1f	cel/cel_odbc: Provide microsecond precision in 'eventtime' column when possible This patch adds microsecond precision when inserting a CEL record into a table with an "eventtime" column of type timestamp, instead of second precision. The documentation (configs/cel_odbc.conf.sample) was already saying that the eventtime column included microseconds precision, but that was not the case. Also, without this patch, if you had a table with an "eventtime" column of type varchar, you had millisecond precision. With this patch, you also get microsecond precision in this case. Review: https://reviewboard.asterisk.org/r/3980 ASTERISK-24283 #close Reported by: Etienne Lessard patches: cel_odbc_time_precision.patch uploaded by Etienne Lessard (License 6394) ........ Merged revisions 427952 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427953 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427954 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ece61f5ed1	chan_pjsip: Add additional log message when an AOR is specified when dialing and it does not exist. ASTERISK-24499 #close Reported by: Rusty Newton ........ Merged revisions 428007 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428008 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	49e63a191d	chan_motif / chan_pjsip: Fix incorrect "No such module" messages when reloading. For chan_motif the direct return value of the underlying config options framework was passed back. This can relay various states which the module loader would not interpet as success. It has been changed so only on errors will it report back an error. For chan_pjsip the code implemented a dummy reload function which always returned an error. This has been removed as all configuration is held within res_pjsip instead. ASTERISK-23651 #close Reported by: Rusty Newton ........ Merged revisions 427981 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427982 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	9d2882d274	res_pjsip: Enforce requirements for session timer minimum expiration period and normal expiration period. This change enforces the requirements in PJSIP for session timer configuration. The minimum expiration period must be 90 seconds or higher and the normal expiration period can not be lower than the minimum expiration period. If either of these were done the code would assert at session setup time. ASTERISK-24336 #close Reported by: Leon Rowland ........ Merged revisions 427978 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427979 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d0523b4b3c	chan_sip: Add support for setting DTLS configuration in the general section. Configuration of DTLS in the general section will be applied to any users or peers. If configuration exists at their level it overrides the general section values. ASTERISK-24128 #close Reported by: Michael K. patches: dtls_default_settings.patch submitted by Michael K. (license 6621) Review: https://reviewboard.asterisk.org/r/3867/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	3268544907	tests/test_cel: Unlock bridge on off nominal paths If the test fails due to memory allocation errors, we may as well attempt to unlock the bridge on the way out. ........ Merged revisions 427927 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	df2090b931	Documentation: Revise explanation of cdr.conf option 'Unanswered' ASTERISK-24279 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4109/ ........ Merged revisions 427901 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427902 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	ba811ae1c3	stun: correct attribute string padding to match rfc When sending the USERNAME attribute in an RTP STUN response, the implementation in append_attr_string passed the actual length, instead of padding it up to a multiple of four bytes as required by the RFC 3489. This change adds separate variables for the string and padded attributed lengths, and performs padding correctly. Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/4139/ ........ Merged revisions 427874 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427875 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427876 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2d9471ab1f	Fix race condition that could result in ARI transfer messages not being sent. From reviewboard: "During blind transfer testing, it was noticed that tests were failing occasionally because the ARI blind transfer event was not being sent. After investigating, I detected a race condition in the blind transfer code. When blind transferring a single channel, the actual transfer operation (i.e. removing the transferee from the bridge and directing them to the proper dialplan location) is queued onto the transferee bridge channel. After queuing the transfer operation, the blind transfer Stasis message is published. At the time of publication, snapshots of the channels and bridge involved are created. The ARI subscriber to the blind transfer Stasis message then attempts to determine if the bridge or any of the involved channels are subscribed to by ARI applications. If so, then the blind transfer message is sent to the applications. The way that the ARI blind transfer message handler works is to first see if the transferer channel is subscribed to. If not, then iterate over all the channel IDs in the bridge snapshot and determine if any of those are subscribed to. In the test we were running, the lone transferee channel was subscribed to, so an ARI event should have been sent to our application. Occasionally, though, the bridge snapshot did not have any channels IDs on it at all. Why? The problem is that since the blind transfer operation is handled by a separate thread, it is possible that the transfer will have completed and the channels removed from the bridge before we publish the blind transfer Stasis message. Since the blind transfer has completed, the bridge on which the transfer occurred no longer has any channels on it, so the resulting bridge snapshot has no channels on it. Through investigation of the code, I found that attended transfers can have this issue too for the case where a transferee is transferred to an application." The fix employed here is to decouple the creation of snapshots for the transfer messages from the publication of the transfer messages. This way, snapshots can be created to reflect what they are at the time of the transfer operation. Review: https://reviewboard.asterisk.org/r/4135 ........ Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427870 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	737b811749	app_confbridge: Play "leader has left" sound even when musiconhold is enabled. Currently if the leader of a conference bridge leaves any participant that has musiconhold enabled will not hear the "leader has left" sound. This is because musiconhold is started and THEN the sound is played. This change makes it so that the sound is played and THEN musiconhold is started. This provides a better experience for users as they may not have known previously why they went back to musiconhold. Review: https://reviewboard.asterisk.org/r/4177/ ........ Merged revisions 427844 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427845 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427846 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2454505d5a	Fix race condition where duplicated requests may be handled by multiple threads. This is the Asterisk 13 version of the patch. The main difference is in the pubsub code since it was completely refactored between Asterisk 12 and 13. Review: https://reviewboard.asterisk.org/r/4175 ........ Merged revisions 427841 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	49b7a1cbaf	res_pjsip_exten_state: PJSIPShowSubscriptionsInbound causes crash When using a non-default sorcery wizard (in this instance realtime) for outbound registrations and after adding in an appropriate call to ast_sorcery_apply_config() (since it is missing) Asterisk will crash after a stack overflow occurs due to the code infinitely recursing. The fix entails removing the outbound registration state dependency from the outbound registration sorcery object and instead keeping an in memory container that can be used to lookup the state when needed. ASTERISK-24514 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4164/ ........ Merged revisions 427814 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427815 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	74e706878b	Stasis: Fix StasisEnd message ordering This change corrects message ordering in cases where a channel-related message can be received after a Stasis/ARI application has received the StasisEnd message. The StasisEnd message was being passed to applications directly without waiting for the channel topic to empty. As a result of this fix, other bugs were also identified and fixed: * StasisStart messages were also being sent directly to apps and are now routed through the stasis message bus properly * Masquerade monitor datastores were being removed at the incorrect time in some cases and were causing StasisEnd messages to not be sent * General refactoring where necessary for the above * Unsubscription on StasisEnd timing changes to prevent additional messages from following the StasisEnd when they shouldn't A channel sanitization function pointer was added to reduce processing and AO2 lookups. Review: https://reviewboard.asterisk.org/r/4163/ ASTERISK-24501 #close Reported by: Matt Jordan ........ Merged revisions 427788 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427789 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cc4c396647	main/rtp_engine: Fix crash when processing more than one RTCP report info block Asterisk - in res_rtp_asterisk - only understands a single RTCP report info block. When the RTCP information was refactored in the RTP Engine to be pushed over the Stasis message bus, I put in the hooks into the engine to handle multiple RTCP report info blocks, in the hope that a future RTP implementation would be able to provide that data. Unfortunately, res_rtp_asterisk has a tendency to "lie": (1) It will send RTCP reports with a reception_report_count greater than 1 (which is pulled directly from the RTCP packet itself, so that part is correct) (2) It will only provide a single report block When the rtp_engine goes to convert this to a JSON blob, hilarity ensues as it looks for a report block that doesn't exist. This patch updates the rtp_engine to be a bit more skeptical about what it is presented with. While this could also be fixed in res_rtp_asterisk, this patch prefers to fix it in the engine for two reasons: (1) The engine is designed to work with multiple RTP implementation, and hence having it be more robust is a good thing (tm) (2) res_rtp_asterisk's handling of RTCP information is "fun". It should report the correct reception_report_count; ideally it should also be giving us all of the blocks - but it is definitely not designed to do that. Going down that road is a non-trivial effort. Review: https://reviewboard.asterisk.org/r/4158/ ASTERISK-24489 #close Reported by: Gregory Malsack Tested by: Gregory Malsack ASTERISK-24498 #close Reported by: Beppo Mazzucato Tested by: Beppo Maazucato ........ Merged revisions 427762 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427763 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ec1a7654f3	Fix leak in AMI Action Bridge Add missing reference cleanup for newly created bridge. ASTERISK-24281 Reported by: Stefan Engström Review: https://reviewboard.asterisk.org/r/4154/ ........ Merged revisions 427736 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427737 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	dbb8f0a935	pbx: Fix off-nominal case where a freed extension may still be used. If during the operation of adding an extension a priority is added but fails it is possible for the extension to be freed but still exist in the PBX core. If this occurs subsequent lookups may try to access the extension and end up in freed memory. This change removes the extension from the PBX core when the priority addition fails and then frees the extension. ASTERISK-24444 #close Reported by: Leandro Dardini Review: https://reviewboard.asterisk.org/r/4162/ ........ Merged revisions 427709 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427710 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427711 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	9f89b83269	Fix compiler error when using ./configure --enable-dev-mode --enable-coverage When DONT_OPTIMIZE is enabled with dev-mode, it causes a shadow compilation to be done with output to /dev/null. This can cause errors with coverage when GCC attempts to write to /dev/null.gcno. This change disables coverage for the shadow compilation. ASTERISK-24502 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4151/ ........ Merged revisions 427682 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427683 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427684 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	21c41e4542	manager: Fix HTTP connection reference leaks. Fix reference leak that happens if (session && !blastaway). ASTERISK-24505 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4153/ ........ Merged revisions 427641 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427642 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427643 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f4392c4b6d	channels/chan_mgcp: Fix regression which causes gateways to be skipped In r227276, a while loop was turned into a for loop. Unfortunately, a portion of the while loop was left in the code such that, when a static gateway is encountered in the list of MGCP gateways, the next gateway would be skipped. At best, we would simply flip past a gateway; at worst, this could lead to a crash. ASTERISK-24500 #close Reported by: Xavier Hienne patches: chan_mgcp.patch uploaded by Xavier Hienne (License 6657) ........ Merged revisions 427613 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427614 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427615 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d773f9d03e	addons/chan_mobile: Increase buffer size of UCS2 encoded SMS messages When UCS2 character encoding is used, one symbol in national language can be expanded to 4 bytes. The current buffer used for receiving message in do_monitor_phone is 256 bytes, which is not large enough for incoming messages. For example: * AT+CMGR phone response prefix '+CMGR: "REC UNREAD","+7*********",,"14/10/29,13:31:39+12"\r\n' - 60 bytes SMS body with UCS2 encoding (max) - 280 bytes * AT+CMGR phone response suffix '\r\n\r\nOK\r\n' - 8 bytes * Terminating null character - 1 byte This results in a needed buffer size of 349 bytes. Hence, this patch opts for a 350 byte buffer. ASTERISK-24468 #close Reported by: Dmitriy Bubnov patches: chan_mobile-1_8.diff uploaded by Dmitriy Bubnov (License 6651) chan_mobile-trunk.diff uploaded by Dmitry Bubnov (License 6651) ........ Merged revisions 427607 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427610 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427611 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	08d773532b	app_voicemail: Fix enhancement that allowed multiple recipients in To: header An issue existed in r420577, which added multiple recipients to voicemail emails. The patch, when looking at the intended recipients, looked ahead for the '\|' character inside a while loop which already had pulled out the appropriate field parsing on the '\|' character. This would cause it to skip the recipients. This patch fixes it such that it relies completely on the while loop to parse through the e-mail fields. Note that the original author of the patch looked at this fix and approved it. ASTERISK-24250 #close Reported by: abelbeck patches: voicemail-420577-to-comma-fix.diff uploaded by abelbeck (License 5903) ........ Merged revisions 427585 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9a1ab5d548	bridge_native_rtp: Fix T.38 issues with remote bridges After r425242 the fax/sip/directmedia_reinvite_t38 test started failing due to the surviving channel not being re-INVITEd back from T.38 to audio. This patch fixes that bug - a deeper explanation of what happened follows. When two RTP channels are in a native bridge, the bridging layer will investigate each via the get_rtp_info glue callback. This callback returns the native bridge preference of the channel at that moment in time (that part is key). At different points during the bridging, the native bridging layer will inform the RTP capable channels of the status of the bridge via the update_peer glue callback. In a T.38 scenario with audio direct media, the sequence of events will often look like the following: * SIP/A and SIP/B both have audio and enter a native bridge. * Asterisk re-INVITEs audio between SIP/A and SIP/B directly (via an update_peer callback). * SIP/A sends a re-INVITE to T.38, which causes Asterisk to send a re-INVITE to T.38 to SIP/B. Assuming everyone 200 OKs the process, the UDPTL stack receives UDPTL packets in Asterisk from both endpoints. From the perspective of the channels, we are now in a local bridge for T.38, even though we are technically still in a remote bridge in bridge_native_rtp. (YAY!) * When one side hangs up, bridge_native_rtp is told to stop bridging. It then re-evaluates the channels and asks them how they are bridged - and since T.38 is enabled, they reply with a Local bridge (which is correct), but is wrong because the audio portion is still technically in a remote bridge. * Asterisk releases the surviving channel, whose audio is not re-INVITED back to Asterisk as bridge_native_rtp incorrectly assumes that it was in a local bridge. Ironically, prior to r425242, this used to work mostly due to a fluke in the bridging layer. The purpose of the get_rtp_info callback shouldn't be modified: it should tell the bridging layer what kind of bridge the channel prefers at that moment in time. If you have T.38 enabled, that must be a local bridge, as the UDPTPL stack must be in the media path. As such, this patch does not modify that part of the code. However, we have to tell the channels to re-evaluate themselves when they come out of a native bridge, since we can no longer trust the get_rtp_info callbacks when the native bridge is being stopped. Something else may have changed in the channels, and they may now be lying to us. As such, this patch makes it so that we unilaterally tell the channels that they are no longer bridged via the update_peer callback. This is actually what the channels expect anyway: code in both chan_sip and chan_pjsip's callbacks look at the T.38 state and - if they were in T.38 - send a re-INVITE to get the audio back to Asterisk. Review: https://reviewboard.asterisk.org/r/4157/ ........ Merged revisions 427582 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427583 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	d4fd0774f4	chan_console: Fix reference leaks to pvt. Fix a bunch of calls to get_active_pvt where the reference is never released. ASTERISK-24504 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4152/ ........ Merged revisions 427554 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427555 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427557 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	7571bae5ab	app_agent_pool: Made agent alert interruptable by DTMF. Made agent able to interrupt the alerting beep playback with DTMF. Any digit can interrupt if the call does not need to be acknowledged. Only the first digit of the acknowledgement can interrupt if the call needs to be acknowledged. The agent interrupting the alerting playback builds on the ASTERISK-24447 patch because it knows what digit interrupted the playback and needs to be able to pass that digit to the DTMF hook digit collection code. ASTERISK-24257 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4123/ ........ Merged revisions 427508 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427512 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a68baad74f	Bridge DTMF hooks: Made audio pass from the bridge while waiting for more matching digits. * Made collecting DTMF digits for the DTMF feature hooks pass frames from the bridge. * Made collecting DTMF digits possible by other bridge hooks if there is a need. ASTERISK-24447 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4123/ ........ Merged revisions 427493 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427494 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	47074f4bfd	res_pjsip: Ensure in-dialog responses have an endpoint associated. When handling incoming messages we determine if it is associated with a dialog. If so we use that to determine what serializer and endpoint to use for the message. Previously this would pass the endpoint to the endpoint lookup module to actually place the endpoint completely on the message. For in-dialog responses, however, this did not occur as dialog processing took over and the endpoint lookup did not occur. This change just places the endpoint in the expected spot immediately instead of relying on the endpoint lookup module. In-dialog responses thus have the expected endpoint. AST-1459 #close Review: https://reviewboard.asterisk.org/r/4146/ ........ Merged revisions 427490 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427491 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	4d80f223af	main/file.c: fix possible extra ast_module_unref to format modules. fn_wrapper only adds a reference to the format's module if the file was able to be opened. If not this causes an unmatched ast_module_unref in filestream_destructor. Move ast_module_ref to get_stream. ASTERISK-24492 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4149/ ........ Merged revisions 427464 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427465 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427466 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	c46664305a	res_hep: fix major leak that occurs when config is missing or enabled=no. Add missing unreference in hepv3_send_packet. ASTERISK-24491 #close Reported by: Zane Conkle Tested by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4150/ ........ Merged revisions 427400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427405 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	7e2369310c	Fix unintential memory retention in stringfields. * Fix missing / unreachable calls to __ast_string_field_release_active. * Reset pool->used to zero when the current pool->active reaches zero. ASTERISK-24307 #close Reported by: Etienne Lessard Tested by: ibercom, Etienne Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........ Merged revisions 427380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427381 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427382 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	362dde2229	test_strings: Remove string tests that exercise asserts. Since unit tests are run with DO_CRASH, those tests were causing the test to fail. Tested-by: George Joseph ........ Merged revisions 427354 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427355 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427356 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	69f29e627f	Make the disable_tcp_switch PJSIP system object enabled by default. Testing has shown repeatedly that PJSIP's default behavior of switching automatically to TCP for large messages can cause issues. The most common issues are that devices that we are communicating with do not handle the switch to TCP gracefully, thus causing situations such as broken calls or broken subscriptions. Now, in order to have this behavior happen, you must opt into it. The sample file has been updated to warn that enabling the TCP switch behavior may cause issues for you, so use at your own risk. ........ Merged revisions 427334 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	b06078880b	res_pjsip_multihomed: Add logging during startup to aid debugging if local DNS is misbehaving. This change adds a bit of logging so if the local DNS is misbehaving it is easier to track down what is going on and where Asterisk may be hanging. ASTERISK-24438 #close Reported by: Melissa Shepherd Review: https://reviewboard.asterisk.org/r/4148/ ........ Merged revisions 427300 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427303 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d5de94201e	config: Make text_file_save and 'dialplan save' escape semicolons in values. When a config file is read, an unescaped semicolon signals comments which are stripped from the value before it's stored. Escaped semicolons are then unescaped and become part of the value. Both of these behaviors are normal and expected. When the config is serialized either by 'dialplan save' or AMI/UpdateConfig however, the now unescaped semicolons are written as-is. If you actually reload the file just saved, the unescaped semicolons are now treated as start of comments. Since true comments are stripped on read, any semicolons in ast_variable.value must have been escaped originally. This patch re-escapes semicolons in ast_variable.values before they're written to file either by 'dialplan save' or config/ast_config_text_file_save which is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting issues nearby in pbx_config.c Tested-by: George Joseph ASTERISK-20127 #close Review: https://reviewboard.asterisk.org/r/4132/ ........ Merged revisions 427275 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427276 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	c77a71ad2f	res_pjsip: Apply the 'user_eq_phone' setting to the To header as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	5e43d68717	res_pjsip: Allow + at the beginning of a phone number when user_eq_phone is enabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

... 2 3 4 5 6 ...

26135 Commits (965777ccfc61067d1dc79f79c12d52f794fa3983) All Branches Search

26135 Commits (965777ccfc61067d1dc79f79c12d52f794fa3983)

All Branches