asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	e910dbab90	build: Execute ldconfig to build cache. On some platforms a multiarch approach is used for libraries. The build system does not take this into account and still places libraries into the lib directory if no --libdir is specified to configure. On initial startup this results in libasteriskssl.so not being found, as it is not in the multiarch lib directory. This change does the minimally invasive thing and executes ldconfig so that the libraries in the lib directory are found and their location cached. By doing so Asterisk starts up fine. ASTERISK-26705 Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519	8 years ago
Joshua Colp	24c2ef2418	Merge "http: Ensure capath is defined on all http creations" into 13	8 years ago
zuul	6b62ab7776	Merge "res_pjsip_pubsub: Correctly implement persisted subscriptions" into 13	8 years ago
zuul	ec0aefc822	Merge "pjsip_distributor.c: Fix off-nominal tdata ref leak." into 13	8 years ago
Joshua Elson	0fc27fa364	http: Ensure capath is defined on all http creations ASTERISK-26794 #close Change-Id: I9cbc3b6b6a8aab590f5ccde9c262a98e4d5253a1	8 years ago
George Joseph	be77b845d9	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	8 years ago
Sean Bright	73133d5354	res_rtp_asterisk: Use PJ_ICE_MAX_CAND instead of hard-coding 16 pjsip limits the total number of ICE candidates to PJ_ICE_MAX_CAND, which is a compile-time constant. Instead of hard-coding 16 when we enumerate local interfaces, use PJ_ICE_MAX_CAND so that we can potentially collect more interfaces if the compile time options are changed. Tangentially related to ASTERISK~24464 Change-Id: I1b85509e39e33b1fed63c86261fc229ba14bbabd	8 years ago
zuul	c5b48fbe77	Merge "libasteriskssl: do nothing with OpenSSL >= 1.1" into 13	8 years ago
zuul	07163005ad	Merge "tcptls: use TLS_client_method with OpenSSL 1.1" into 13	8 years ago
zuul	258b808dce	Merge "openssl 1.1 support: use OPENSSL_VERSION_NUMBER" into 13	8 years ago
zuul	4551525fa9	Merge "app_voicemail: Allow 'Comedian Mail' branding to be overriden" into 13	8 years ago
zuul	730bc7e972	Merge "app_voicemail: VoiceMailPlayMsg did not play database stored messages" into 13	8 years ago
Tzafrir Cohen	99b40e72ae	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	8 years ago
Sean Bright	4c31e03e80	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	8 years ago
zuul	500b02cb82	Merge "app_record: Add option to prevent silence from being truncated" into 13	8 years ago
Tzafrir Cohen	e97e50b68b	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	8 years ago
Tzafrir Cohen	0d555f0d81	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	8 years ago
zuul	0d6c99e715	Merge "cli: Fix various CLI documentation and completion issues" into 13	8 years ago
zuul	bc2104819c	Merge "channel: Protect flags in ast_waitfor_nandfds operation." into 13	8 years ago
rrittgarn	9d34df9a5e	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	8 years ago
Sean Bright	f99e5f4de4	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	8 years ago
zuul	e0c52c5ec2	Merge "res_pjsip.c: Fix inconsistency between warning and action." into 13	8 years ago
zuul	6958241b3f	Merge "core: Cleanup some channel snapshot staging anomalies." into 13	8 years ago
Sean Bright	ea8a610776	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/asterisk: Correct and extend completions for 'core show file version.' * 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	8 years ago
zuul	c38cd504ad	Merge "chan_pjsip: Multidomain endpoint finding on call" into 13	8 years ago
Norbert Varga	17030100ca	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	8 years ago
Joshua Colp	18f1b52601	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	8 years ago
Richard Mudgett	a46a21642e	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	8 years ago
Richard Mudgett	67b21dc63a	pjsip_distributor.c: Fix off-nominal tdata ref leak. Change-Id: I571f371d0956a8039b197b4dbd8af6b18843598d	8 years ago
Sean Bright	8936568515	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	8 years ago
Richard Mudgett	2817f87d27	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	8 years ago
George Joseph	c7fcc4468f	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	8 years ago
Mark Michelson	cbc23c31cf	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	8 years ago
Joshua Colp	2a3df5aa22	Merge "srv: Fix crash when ast_srv_lookup is used and 0 records are returned." into 13	8 years ago
zuul	3da771bc3a	Merge "res_stasis_device_state: Protect the adding/removing of subscriptions." into 13	8 years ago
Joshua Colp	7e14e086cf	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	8 years ago
Joshua Colp	7b39d6901a	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	8 years ago
Richard Mudgett	c384dfd6b0	res_pjsip: Fix some off nominal tdata leaks. Change-Id: I243a4be5e7fbfe604923764969c4ee04eee89b9d	8 years ago
zuul	431923feb6	Merge "Update qualifies when AOR configuration changes." into 13	8 years ago
George Joseph	95e33a1f31	Merge "channel.c: Fix unbalanced read queue deadlocking local channels." into 13	8 years ago
zuul	ba201f71af	Merge "res_agi: Prevent an AGI from eating frames it should not. (Re-do)" into 13	8 years ago
zuul	b6eccdbef2	Merge "Add reload options to CLI/AMI stale object commands." into 13	8 years ago
zuul	0bb5c58cba	Merge "Frame deferral: Revert API refactoring." into 13	8 years ago
Sean Bright	70aff89e5d	res_odbc: Remove deprecated settings from sample configuration file ASTERISK-26704 #close Reported by: Anthony Messina Change-Id: I976a1f94cf79c5f31e76174c61f5c6a65fd6354f	8 years ago
Joshua Colp	60f2e96d62	Merge "audiohooks: Muting a hook can mute underlying frames" into 13	8 years ago
Joshua Colp	4e18f9afac	Merge "res_pjsip: Handle invocation of callback on outgoing request when error occurs." into 13	8 years ago
Sean Bright	3aee199913	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	8 years ago
zuul	e4f742de15	Merge "res_rtp_asterisk: Swap byte-order when sending signed linear" into 13	8 years ago
Mark Michelson	6492e91392	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	8 years ago
Richard Mudgett	43f0ff4b69	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	8 years ago

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