asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	ad85e54fd9	res_pjsip_t38: Fix T.38 failure when peer reinvites immediately. If a remote endpoint reinvites to T.38 immediately the state machine will go into a peer reinvite state. If a T.38 capable application (such as ReceiveFax) queries it will receive this state. Normally the application will then indicate so that the channel driver will queue up the T.38 offer previously received. Once it receives this offer the application will act normally and negotiate. The res_pjsip_t38 module incorrectly partially squashed this indication. This would cause the application to think the request had failed when in reality it had actually worked. This change makes it so that no T.38 control frames (or indications) are squashed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	89617370ec	res_pjsip_config_wizard: Allow streamlined config of common pjsip scenarios res_pjsip_config_wizard ------------------ * This is a new module that adds streamlined configuration capability for chan_pjsip. It's targetted at users who have lots of basic configuration scenarios like 'phone' or 'agent' or 'trunk'. Additional information can be found in the sample configuration file at config/samples/pjsip_wizard.conf.sample. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4190/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	b85f79c0c1	Activate persistent subscriptions when they are recreated. Prior to this change, recreating persistent subscriptions would create the subscription but would not activate it. This led to subscriptions being listed in the "NULL" state by diagnostics and not sending NOTIFYs when expected. Review: https://reviewboard.asterisk.org/r/4261 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	8c019b1a6b	res/res_agi: Make Verbose message for 'stream file' match other playbacks The Verbose message displayed when a file is played back via 'stream file' was formatted differently than other playbacks: * It didn't include the channel name * It didn't include the channel language It does, however, include the playback offset as well as any escape digits. That information was kept; however, this patch updates the formatting to more closely match the Verbose messages displayed when a file is played back by 'control stream file', Playback, ControlPlayback, or any other file playback operation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	49386cf568	Fix crash for sorcery misconfigs res_pjsip_outbound_publish was missing the CHECK_PJSIP_MODULE_LOADED() call in load_module, and would crash with a segfault if res_pjsip declined to load. Review: https://reviewboard.asterisk.org/r/4258/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	3b0c40f337	PJSIP: Allow use of 'inactive' streams for hold This allows use of the 'inactive' stream direction identifier to be used for hold where 'sendonly' is normally used. Some Seimens phones use 'inactive' and this change allows music on hold to operate properly. Review: https://reviewboard.asterisk.org/r/4252/ Reported by: Steve Pitts ........ Merged revisions 429432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	15af40180a	Sorcery: Log when old config remains in use This adds a log message notifying the user that a stale configuration is in place upon reload when a config object fails to load. This situation can end up causing confusion when the object failed to load but exists from a previous config load especially when the old config is significantly different from the new config. Review: https://reviewboard.asterisk.org/r/4250/ Reported by: Thomas Thompson ........ Merged revisions 429429 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	0c9fbb449f	res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress. Given the scenario where a PJSIP channel is in a native RTP bridge with direct media and the channel is then hung up the code will currently re-INVITE the channel back to Asterisk and send a BYE at the same time. Many SIP implementations dislike this greatly. This change makes it so that if a re-INVITE transaction is in progress the BYE is queued to occur after the completion of the transaction (be it through normal means or a timeout). Review: https://reviewboard.asterisk.org/r/4248/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	61fe4f10d2	res_pjsip_session: Fix issue where a declined media stream in a re-INVITE would fail SDP negotiation. In the past the SDP negotiation within res_pjsip_session was made more tolerant of certain situations. The only case where SDP negotiation will fail is when a major error occurs during negotiation. Receiving an already declined media stream is not considered a major error. When producing the local SDP the logic took this into account so on the initial INVITE the declined media stream did not cause an SDP negotiation failure. Unfortunately the logic for handling media streams with a handler did not mirror this logic and considered an already declined media stream an error and thus failed the SDP negotiation. This change makes the logic between both situations match so only under major errors will the SDP negotiation fail. ASTERISK-24607 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4254/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	fb768ec33a	res_http_websocket: Fix crash due to double freeing memory when receiving a payload length of zero. Frames with a payload length of 0 were incorrectly handled in res_http_websocket. Provided a frame with a payload had been received prior it was possible for a double free to occur. The realloc operation would succeed (thus freeing the payload) but be treated as an error. When the session was then torn down the payload would be freed again causing a crash. The read function now takes this into account. This change also fixes assumptions made by users of res_http_websocket. There is no guarantee that a frame received from it will be NULL terminated. ASTERISK-24472 #close Reported by: Badalian Vyacheslav Review: https://reviewboard.asterisk.org/r/4220/ Review: https://reviewboard.asterisk.org/r/4219/ ........ Merged revisions 429270 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429272 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	a220a08777	PJSIP: Fix assert on initial mass qualify This fixes the MWI test regressions caused by r429127 and ensures that contacts have non-zero qualify_frequency before attempting scheduling. ........ Merged revisions 429245 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	2f21f85c37	ARI/AMI: Include language in standard channel snapshot output The channel "language" was already part of a channel snapshot, however is was not sent out over AMI or ARI. This patch makes it so the channel "language" is included in the appropriate AMI or ARI events. ASTERISK-24553 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4245/ ........ Merged revisions 429204 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	525c823b4b	Direct Media calls within private network sometimes get one way audio When endpoints with direct_media enabled, behind a firewall (Asterisk on a separate network) and were bridged sometimes Asterisk would send the ip address of the firewall in the sdp to one of the phones in the reinvite resulting in one way audio. When sending the reinvite Asterisk will retrieve the media address from the associated rtp instance, but if frames were being read this can be overwritten with another address (in this case the firewall's). This patch ensures that Asterisk uses the original device address when using direct media. ASTERISK-24563 Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4216/ ........ Merged revisions 429195 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	664067e318	res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard When using a non-default sorcery wizard (in this instance realtime) for outbound publishes Asterisk will crash after a stack overflow occurs due to the code infinitely recursing. The fix entails removing the outbound publish state dependency from the outbound publish sorcery object and instead keeping an in memory container that can be used to lookup the state when needed. ASTERISK-24514 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4178/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	74b032bb03	ari: Add support for specifying an originator channel when originating. If an originator channel is specified when originating a channel the linked ID of it will be applied to the newly originated outgoing channel. This allows an association to be made between the two so it is known that the originator has dialed the originated channel. ASTERISK-24552 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4243/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	64581d894d	PJSIP: Stagger outbound qualifies This change staggers initiation of outbound qualify (OPTIONS) attempts to reduce instantaneous server load and prevent network congestion. Review: https://reviewboard.asterisk.org/r/4246/ ASTERISK-24342 #close Reported by: Richard Mudgett ........ Merged revisions 429127 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429128 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	1712694817	Fix a crash that would occur when receiving a 491 response to a reinvite. The reviewboard description does a fine job of summarizing this, so here it is: A reporter discovered that Asterisk would crash when attempting to retransmit a reinvite that had previously received a 491 response. The crash occurred because a pjsip_tx_data structure was being saved for reuse, but its reference count was not being increased. The result was that the pjsip_tx_data was being freed before we were actually done with it. When we attempted to re-use the structure when re-sending the reinvite, Asterisk would crash. The fix implemented here is not to try holding onto the pjsip_tx_data at all. Instead, when we reschedule sending the reinvite, we create a brand new pjsip_tx_data and send that instead. Because of this change, there is no need for an ast_sip_session_delayed_request structure to have a pjsip_tx_data on it any more. So any code referencing its use has been removed. When this initial fix was introduced, I encountered a second crash when processing a subsequent 200 OK on a rescheduled reinvite. The reason was that when rescheduling the reinvite, we gave the wrong location for a response callback. This has been fixed in this patch as well. ASTERISK-24556 #close Reported by Abhay Gupta Review: https://reviewboard.asterisk.org/r/4233 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	93b1df3bf6	Add new AMI and ARI events for connected line changes on a channel. The AMI event is called NewConnectedLine and the ARI event is called ChannelConnectedLine. ASTERISK-24554 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4231 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	55c9a46abd	Stasis: Fix StasisStart/End order and missing events This corrects several bugs that currently exist in the stasis application code. * After a masquerade, the resulting channels have channel topics that do not match their uniqueids ** Masquerades now swap channel topics appropriately * StasisStart and StasisEnd messages are leaked to observer applications due to being published on channel topics ** StasisStart and StasisEnd publishing is now properly restricted to controlling apps via app topics * Race conditions exist where StasisStart and StasisEnd messages due to a masquerade may be received out of order due to being published on different topics ** These messages are now published directly on the app topic so this is now a non-issue * StasisEnds are sometimes missing when sent due to masquerades and bridge swaps into and out of Stasis() This was due to StasisEnd processing adjusting message-sent flags after Stasis() had already exited and Stasis() had been re-entered This was corrected by adjusting these flags prior to sending the message while the initial Stasis() application was still shutting down Review: https://reviewboard.asterisk.org/r/4213/ ASTERISK-24537 #close Reported by: Matt DiMeo ........ Merged revisions 429061 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7fd1227202	res/res_monitor: Reset in/out sample counts on Monitor start When repeatedly starting/stopping a Monitor on a channel, the accumulated in/out sample counts are never reset to 0. This can cause inadvertent jumps in the recordings, as the code in the channel core will determine incorrectly that a jump in the recorded file position should occur. Setting the sample counts to 0 simply reflects the initial state a Monitor should be in when it is started, as this is the initial count that would be on the channels at that time. ASTERISK-24573 #close Reported by: Nuno Borges patches: 24573.patch uploaded by Nuno Borges (License 6116) ........ Merged revisions 429031 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429032 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	a551851e5b	res_pjsip_refer: Fix issue where native bridge may not occur upon completion of a transfer. There are two methods within res_pjsip_refer for keeping track of the state of a transfer. The first is a framehook which looks at frames passing by to determine the state. The second subscribes to know when the channel joins a bridge. In the case when the channel joins the bridge the framehook is NOT removed and this prevents the native RTP bridging technology from getting used. This change gets the channel and if it still exists remove the framehook. Review: https://reviewboard.asterisk.org/r/4218/ ........ Merged revisions 428760 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	7472c661ca	res_pjsip_endpoint_identifier_ip: Add 'show identify(ies)' cli commands While troubleshooting other things I realized there were no pjsip cli commands for identify. This patch adds them. It also also fixes a reference leak when a 'show endpoint' displayed identifies and properly sets the return code if load_module can't allocate a cli formatter structure. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4212/ ........ Merged revisions 428725 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d79c68d3fb	main/stasis: Allow subscriptions to use a threadpool for message delivery Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	2846d23688	sorcery: Make is_object_field_registered handle field names that are regexes. As a result of https://reviewboard.asterisk.org/r/3305, res_sorcery_realtime was tossing database fields that didn't have an exact match to a sorcery registered field. This broke the ability to use regexes as field names which manifested itself as a failure of res_pjsip_phoneprov_provider which uses this capability. It also broke handling of fields that start with '@' in realtime but I don't think anyone noticed. This patch does the following... * Modifies ast_sorcery_fields_register to pre-compile the name regex. * Modifies ast_sorcery_is_object_field_registered to test the regex if it exists instead of doing an exact strcmp. * Modifies res_pjsip_phoneprov_provider with a few tweaks to get it to work with realtime. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4185/ ........ Merged revisions 428543 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	ddc3f4846f	PJSIP ACLs: Fix ACLs not loading on startup and apply/acl issues on contact The biggest problem this patch fixes is that ACLs weren't previously being loaded when the res_pjsip_acl module was loaded. Yikes. In addition, the ACL options contact_permit and contact_acl were effectively interpreted as contact_deny and this patch fixes that as well. AST-1418 #close Reported by: Thomas Thompson Review: https://reviewboard.asterisk.org/r/4120/ ASTERISK-24531 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4171/ ........ Merged revisions 428333 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e7f16827cb	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	524588c345	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	a0d9eab389	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	87c2e9e238	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	8648bc1f6f	parking_tests.c: Add missing newline on a unit test message. ........ Merged revisions 428168 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428169 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	cbba061f13	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	1536b0ecb6	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	77c57f2b1a	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	721ae89155	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427815 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	5013659972	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	248c592292	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	1c255e711e	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	27dc7e47d7	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	0d0131bf34	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	3e29fa3686	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b586e7f0b2	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d5309929be	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	b3b93a7c15	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	b4ee155c62	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	c3b1d0df0d	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	fed9d0deb0	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	0ca681a414	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Malcolm Davenport	0bbb351655	ASTERISK-24323, fix bug in documentation of AGI STREAM FILE CONTROL git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b23580afe6	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f3fbcc550e	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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2758 Commits (f26d4618eb7dff4e6c8e8dfd31765bb331bdf2ba)