asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	507c50914a	res_pjsip_pubsub: Correctly implement persisted subscriptions This patch fixes 2 original issues and more that those 2 exposed. * When we send a NOTIFY, and the client either doesn't respond or responds with a non OK, pjproject only calls our pubsub_on_evsub_state callback, no others. Since pubsub_on_evsub_state (which does the sub_tree cleanup) does not expect to be called back without the other callbacks being called first, it just returns leaving the sub_tree orphaned. Now pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE which is what pjproject will set to tell us that it was the transaction that timed out or failed and not the subscription itself timing our or being terminated by the client. If is TSX_STATE, pubsub_on_evsub_state now does the proper cleanup regardless of the state of the subscription. * When a client renews a subscription, we don't update the persisted subscription with the new expires timestamp. This causes subscription_persistence_recreate to prune the subscription if/when asterisk restarts. Now, pubsub_on_rx_refresh calls subscription_persistence_update to apply the new expires timestamp. This exposed other issues however... * When creating a dialog from rdata (which sub_persistence_recreate does from the packet buffer) there must NOT be a tag on the To header (which there will be when a client refreshes a subscription). If there is one, pjsip_dlg_create_uas will fail. To address this, subscription_persistence_update now accepts a flag that indicates that the original packet buffer must not be updated. New subscribes don't set the flag and renews do. This makes sure that when the rdata is recreated on asterisk startup, it's done from the original subscribe packet which won't have the tag on To. * When creating a dialog from rdata, we were setting the dialog's remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq. When the client tried to resubscribe after a restart with the correct cseq, we'd reject the request with an Invalid CSeq error. * The acts of creating a dialog and evsub by themselves when recreating a subscription does NOT restart pjproject's subscription timer. The result was that even if we did correctly recreate the subscription, we never removed it if the client happened to go away or send a non-OK response to a NOTIFY. However, there is no pjproject function exposed to just set the timer on an evsub that wasn't created by an incoming subscribe request. To address this, we create our own timer using ast_sip_schedule_task. This timer is used only for re-establishing subscriptions after a restart. An earlier approach was to add support for setting pjproject's timer (via a pjproject patch) and while that patch is still included here, we don't use that call at the moment. While addressing these issues, additional debugging was added and some existing messages made more useful. A few formatting changes were also made to 'pjsip show scheduled tasks' to make displaying the subscription timers a little more friendly. ASTERISK-26696 ASTERISK-26756 Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e	9 years ago
Joshua Colp	5b308afa60	Merge "libasteriskssl: do nothing with OpenSSL >= 1.1" into 14	9 years ago
Joshua Colp	a0553d8bda	Merge "tcptls: use TLS_client_method with OpenSSL 1.1" into 14	9 years ago
zuul	22339bdaa7	Merge "openssl 1.1 support: use OPENSSL_VERSION_NUMBER" into 14	9 years ago
zuul	6db4ac4438	Merge "app_voicemail: Allow 'Comedian Mail' branding to be overriden" into 14	9 years ago
zuul	a65a73a1ef	Merge "app_voicemail: VoiceMailPlayMsg did not play database stored messages" into 14	9 years ago
Sean Bright	23ee13dcd1	app_voicemail: Allow 'Comedian Mail' branding to be overriden Original patch by John Covert, slight modifications by me. ASTERISK-17428 #close Reported by: John Covert Patches: app_voicemail.c.patch (license #5512) patch uploaded by John Covert Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6	9 years ago
zuul	40dbbaebe3	Merge "app_record: Add option to prevent silence from being truncated" into 14	9 years ago
Tzafrir Cohen	fabccd8831	libasteriskssl: do nothing with OpenSSL >= 1.1 OpenSSL 1.1 requires no explicit initialization. The hacks in the library are not needed. They also happen to fail running Asterisk. ASTERISK-26109 #close Change-Id: I3b3efd5d80234a4c45a8ee58dcfe25b15d9ad100	9 years ago
Tzafrir Cohen	ab8b7b9ac1	tcptls: use TLS_client_method with OpenSSL 1.1 OpenSSL 1.1 introduced TLS_client_method() and deprecated the previous version-specific methods (such as TLSv1_client_method(). Other than being simpler to use and more correct (gain support for TLS newer that TLS1, in our case), the older ones produce a deprecation warning that fails the build in dev-mode. ASTERISK-26109 #close Change-Id: I257b1c8afd09dcb0d96cda3a41cb9f7a15d0ba07	9 years ago
Tzafrir Cohen	c572e17266	openssl 1.1 support: use OPENSSL_VERSION_NUMBER Use OPENSSL_VERSION_NUMBER instead of OPENSSL_API_COMPAT to detect the openssl 1.1 API. ASTERISK-26109 #close Change-Id: I4e448f55ef516aedf6ad154037c35577a421a458	9 years ago
zuul	18ec2919cf	Merge "cli: Fix various CLI documentation and completion issues" into 14	9 years ago
zuul	a505e42793	Merge "channel: Protect flags in ast_waitfor_nandfds operation." into 14	9 years ago
rrittgarn	9296a728a9	app_voicemail: VoiceMailPlayMsg did not play database stored messages When attempting to use VoiceMailPlayMsg with a realtime data backend the message is located, but never retrieved. This patch adds the required RETRIEVE and DISPOSE calls that will fetch the message from the database (and IMAP storage as well for that matter). Also, removed extraneous make_file call. ASTERISK-26723 #close Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c	9 years ago
Sean Bright	fa8a1eff2f	app_record: Add option to prevent silence from being truncated When using Record() with the silence detection feature, the stream is written out to the given file. However, if only 'silence' is detected, this file is then truncated to the first second of the recording. This patch adds the 'u' option to Record() to override that behavior. ASTERISK-18286 #close Reported by: var Patches: app_record-1.8.7.1.diff (license #6184) patch uploaded by var Change-Id: Ia1cd163483235efe2db05e52f39054288553b957	9 years ago
zuul	6743541540	Merge "res_pjsip.c: Fix inconsistency between warning and action." into 14	9 years ago
zuul	4ce60b8419	Merge "core: Cleanup some channel snapshot staging anomalies." into 14	9 years ago
Sean Bright	dc8dd56684	cli: Fix various CLI documentation and completion issues * app_minivm: Use built-in completion facilities to complete optional arguments. * app_voicemail: Use built-in completion facilities to complete optional arguments. * app_confbridge: Add missing colons after 'Usage' text. * chan_alsa: Use built-in completion facilities to complete optional arguments. * chan_sip: Use built-in completion facilities to complete optional arguments. Add completions for 'load' for 'sip show user', 'sip show peer', and 'sip qualify peer.' * chan_skinny: Correct and extend completions for 'skinny reset' and 'skinny show line.' * func_odbc: Correct completions for 'odbc read' and 'odbc write' * main/astmm: Use built-in completion facilities to complete arguments for 'memory' commands. * main/bridge: Correct completions for 'bridge kick.' * main/ccss: Use built-in completion facilities to complete arguments for 'cc cancel' command. * main/cli: Add 'all' completion for 'channel request hangup.' Correct completions for 'core set debug channel.' Correct completions for 'core show calls.' * main/pbx_app: Remove redundant completions for 'core show applications.' * main/pbx_hangup_handler: Remove unused completions for 'core show hanguphandlers all.' * res_sorcery_memory_cache: Add completion for 'reload' argument of 'sorcery memory cache stale' and properly implement. Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca	9 years ago
zuul	0aecb8759b	Merge "chan_pjsip: Multidomain endpoint finding on call" into 14	9 years ago
Norbert Varga	137b38f946	chan_pjsip: Multidomain endpoint finding on call When PJSIP tries to call an endpoint with a domain (e.g. 1000@test.com), the user part is stripped down as it would be a trunk with a specified user, and only the host part is called as a PJSIP endpoint and can't be found. This is not correct in the case of a multidomain SIP account, so the stripping after the @ sign is done only if the whole endpoint (in multidomain case 1000@test.com) can't be found. ASTERISK-26248 Change-Id: I3a2dd6f57f3bd042df46b961eccd81d31ab202e6	9 years ago
Joshua Colp	0956ca15dc	channel: Protect flags in ast_waitfor_nandfds operation. The ast_waitfor_nandfds operation will manipulate the flags of channels passed in. This was previously done without the channel lock being held. This could result in incorrect values existing for the flags if another thread manipulated the flags at the same time. This change locks the channel during flag manipulation. ASTERISK-26788 Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed	9 years ago
Richard Mudgett	02c43d5597	res_pjsip.c: Fix inconsistency between warning and action. The original return value corresponded to AST_SIP_AUTHENTICATION_CHALLENGE but we have no authenticator registered to create the challenge. Change-Id: I62368180d774b497411b80fbaabd0c80841f8512	9 years ago
Sean Bright	c74af2951b	manager: Restore Originate failure behavior from Asterisk 11 In Asterisk 11, if the 'Originate' AMI command failed to connect the provided Channel while in extension mode, a 'failed' extension would be looked up and run. This was, I believe, unintentionally removed in `51b6c49`. This patch restores that behavior. This also adds an enum for the various 'synchronous' modes in an attempt to make them meaningful. ASTERISK-26115 #close Reported by: Nasir Iqbal Change-Id: I8afbd06725e99610e02adb529137d4800c05345d	9 years ago
Richard Mudgett	f88b598ca2	core: Cleanup some channel snapshot staging anomalies. We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a	9 years ago
George Joseph	abf9561518	configs/samples: Fix placement of 'identify' entry in sorcery.conf The entry for 'identify' was incorrectly placed in the res_pjsip section when it should be in res_pjsip_endpoint_identifier_ip. ASTERISK-26785 #close Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a	9 years ago
Mark Michelson	8a528abe9e	Revert "Update qualifies when AOR configuration changes." This reverts commit `6492e91392`. The change in question was intended to prevent the need to reload in order to update qualifies on contacts when an AOR changes. However, this ended up causing a deadlock instead. Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e	9 years ago
zuul	a49870b8df	Merge "srv: Fix crash when ast_srv_lookup is used and 0 records are returned." into 14	9 years ago
Joshua Colp	861e5d2114	srv: Fix crash when ast_srv_lookup is used and 0 records are returned. When performing an SRV lookup using the ast_srv_lookup function it did not properly handle the situation where 0 records are returned. If this happened it would wrongly assume that at least one record was present. This change fixes the code so it will exit early if an error occurs or if 0 records are returned. ASTERISK-26772 patches: srv_lookup.patch submitted by nappsoft (license 6822) Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351	9 years ago
Joshua Colp	7e20cf95cf	res_stasis_device_state: Protect the adding/removing of subscriptions. The adding and removing of device state subscriptions did not protect fully against simultaneous manipulation. In particular the subscribe case allowed a small window where two subscriptions could be added for the same device state instead of just one. This change makes the code hold the subscriptions lock for the entirety of each operation to ensure that two are not occurring at the same time. ASTERISK-26770 Change-Id: I3e7f8eb9d09de440c9024d2dd52029f6f20e725b	9 years ago
Richard Mudgett	382f1d856f	res_pjsip: Fix some off nominal tdata leaks. Change-Id: I243a4be5e7fbfe604923764969c4ee04eee89b9d	9 years ago
Sebastien Duthil	f9beac8071	res_ari: fix memory leak for channelvars In ari.conf, when setting the option channelvars, every Stasis channel snapshot would create a list of variable/value that would not be freed when the snapshot is freed, resulting in a often-recurring memory leak. ASTERISK-26767 #close Change-Id: Ia37dd9d68063d7f879193df02ede293e5ded716d	9 years ago
Joshua Colp	fc7258b053	Merge "Update qualifies when AOR configuration changes." into 14	9 years ago
zuul	e8105c8eb1	Merge "channel.c: Fix unbalanced read queue deadlocking local channels." into 14	9 years ago
zuul	bea59cbe9a	Merge "res_agi: Prevent an AGI from eating frames it should not. (Re-do)" into 14	9 years ago
zuul	1f3a1998e8	Merge "Add reload options to CLI/AMI stale object commands." into 14	9 years ago
Joshua Colp	ddc92b367b	Merge "Frame deferral: Revert API refactoring." into 14	9 years ago
zuul	304b65586c	Merge "res_odbc: Remove deprecated settings from sample configuration file" into 14	9 years ago
Richard Mudgett	1a0478a6c4	channel.c: Fix unbalanced read queue deadlocking local channels. Using the timerfd timing module can cause channel freezing, lingering, or deadlock issues. The problem is because this is the only timing module that uses an associated alert-pipe. When the alert-pipe becomes unbalanced with respect to the number of frames in the read queue bad things can happen. If the alert-pipe has fewer alerts queued than the read queue then nothing might wake up the thread to handle received frames from the channel driver. For local channels this is the only way to wake up the thread to handle received frames. Being unbalanced in the other direction is less of an issue as it will cause unnecessary reads into the channel driver. ASTERISK-26716 is an example of this deadlock which was indirectly fixed by the change that found the need for this patch. * In channel.c:__ast_queue_frame(): Adding frame lists to the read queue did not add the same number of alerts to the alert-pipe. Correspondingly, when there is an exceptionally long queue event, any removed frames did not also remove the corresponding number of alerts from the alert-pipe. ASTERISK-26632 #close Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6	9 years ago
Richard Mudgett	d23fed8f2a	res_agi: Prevent an AGI from eating frames it should not. (Re-do) A dialplan intercept routine is equivalent to an interrupt routine. As such, the routine must be done quickly and you do not have access to the media stream. These restrictions are necessary because the media stream is the responsibility of some other code and interfering with or delaying that processing is bad. A possible future dialplan processing architecture change may allow the interception routine to run in a different thread from the main thread handling the media and remove the execution time restriction. * Made res_agi.c:run_agi() running an AGI in an interception routine run in DeadAGI mode. No touchy channel frames. ASTERISK-25951 ASTERISK-26343 ASTERISK-26716 Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43	9 years ago
Richard Mudgett	4c4be0e0be	Frame deferral: Revert API refactoring. There are several issues with deferring frames that are caused by the refactoring. 1) The code deferring frames mishandles adding a deferred frame to the deferred queue. As a result the deferred queue can only be one frame long. 2) Deferrable frames can come directly from the channel driver as well as the read queue. These frames need to be added to the deferred queue. 3) Whoever is deferring frames is really only doing the __ast_read() to collect deferred frames and doesn't care about the returned frames except to detect a hangup event. When frame deferral is completed we must make the normal frame processing see the hangup as a frame anyway. As such, there is no need to have varying hangup frame deferral methods. We also need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. That fake hangup is to cause the PBX thread to break out of loops to go execute a new dialplan location. 4) To properly deal with deferrable frames from the channel driver as pointed out by (2) above, means that it is possible to process a dialplan interception routine while frames are deferred because of the AST_CONTROL_READ_ACTION control frame. Deferring frames is not implemented as a re-entrant operation so you could have the unsupported case of two sections of code thinking they have control of the media stream. A worse problem is because of the bad implementation of the AMI PlayDTMF action. It can cause two threads to be deferring frames on the same channel at the same time. (ASTERISK_25940) * Rather than fix all these problems simply revert the API refactoring as there is going to be only autoservice and safe_sleep deferring frames anyway. ASTERISK-26343 ASTERISK-26716 #close Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496	9 years ago
zuul	cd48b8044e	Merge "audiohooks: Muting a hook can mute underlying frames" into 14	9 years ago
Sean Bright	bd0258b0b1	res_odbc: Remove deprecated settings from sample configuration file ASTERISK-26704 #close Reported by: Anthony Messina Change-Id: I976a1f94cf79c5f31e76174c61f5c6a65fd6354f	9 years ago
Joshua Colp	4220378d58	Merge "res_pjsip: Handle invocation of callback on outgoing request when error occurs." into 14	9 years ago
Richard Mudgett	b918bd858f	res_resolver_unbound.c: Fix frequent ref leak caught by excessive ref trap. ASTERISK-26765 Change-Id: I27eb97df7f8d7e624b0b9a61c0fcee4718c86d8d	9 years ago
Sean Bright	6e0d7e653b	audiohooks: Muting a hook can mute underlying frames If an audiohook is placed on a channel that does not require transcoding, muting that hook will cause the underlying frames to be muted as well. The original patch is from David Woolley but I have modified slightly. ASTERISK-21094 #close Reported by: David Woolley Patches: ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded by David Woolley Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed	9 years ago
Joshua Colp	d6502d3cbd	Merge "res_rtp_asterisk: Swap byte-order when sending signed linear" into 14	9 years ago
Mark Michelson	fd6b820b4d	Update qualifies when AOR configuration changes. Prior to this change, qualifies would only update in the following cases: * A reload of res_pjsip.so was issued. * A dynamic contact was re-registered after its AOR's qualify_frequency had been changed This does not work well if you are using realtime for your AORs. You can update your database to have a new qualify_frequency, but the permanent contacts on that AOR will not have their qualifies updated. And the dynamic contacts on that AOR will not have their qualifies updated until the next registration, which could be a long time. This change seeks to fix this problem by making it so that whenever AOR configuration is applied, the contacts pertaining to that AOR have their qualifies updated. Additions from this patch: * AOR sorcery objects now have an apply handler that calls into a newly added function in the OPTIONS code. This causes all contacts associated with that AOR to re-schedule qualifies. * When it is time to qualify a contact, the OPTIONS code checks to see if the AOR can still be retrieved. If not, then qualification is canceled on the contact. Alterations from this patch: * The registrar code no longer updates contact's qualify_frequence and qualify_timeout. There is no point to this since those values already get updated when the AOR changes. * Reloading res_pjsip.so no longer calls the OPTIONS initialization function. Reloading res_pjsip.so results in re-loading AORs, which results in re-scheduling qualifies. Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121	9 years ago
Joshua Colp	dcea8f13ce	res_pjsip: Handle invocation of callback on outgoing request when error occurs. There are some error cases in PJSIP when sending a request that will result in the callback for the request being invoked. The code did not handle this case and assumed on every error case that the callback was not invoked. The code has been changed to check whether the callback has been invoked and if so to absorb the error and treat it as a success. ASTERISK-26679 ASTERISK-26699 Change-Id: I563982ba204da5aa1428989a11c06dd9087fea91	9 years ago
Sean Bright	3273ea7384	res_rtp_asterisk: Swap byte-order when sending signed linear Before Asterisk 13, signed linear was converted into network byte order by a smoother before being sent over the network. We restore this behavior by forcing the creation of a smoother when slinear is in use and setting the appropriate flags so that the byte order conversion is always done. ASTERISK-24858 #close Reported-by: Frankie Chin Change-Id: I868449617d1a7819578f218c8c6b2111ad84f5a9	9 years ago
George Joseph	6cc7ee85df	debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts Forgot to install it with the original patch Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c	9 years ago

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