asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	67a45e0a38	Merge "sorcery: Add setting object type congestion levels."	10 years ago
Joshua Colp	48c2c3b8da	Merge "taskprocessors: Implement high/low water mark alerts."	10 years ago
Joshua Colp	2b840fbc3f	Merge "res_pjsip_session: Use distributor serializer for incoming calls."	10 years ago
Joshua Colp	66e1e0969b	Merge "res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer."	10 years ago
Joshua Colp	2b2ca82e71	Merge "res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions."	10 years ago
Joshua Colp	9acb5e3084	Merge "pjsip_distributor.c: Consistently pick a serializer for messages."	10 years ago
zuul	f5ffcb1b72	Merge "pjsip_distributor.c: Ignore messages until fully booted."	10 years ago
Joshua Colp	d338343dac	cel: Ensure only one dial status per channel exists. CEL wrongly assumed that a channel would only have a single dial event on it. This is incorrect. Particularly in a queue each call attempt to a member will result in a dial event, adding a new dial status in CEL without removing the old one. This would cause the container to grow with only one dial status being removed when the channel went away. The other dial status entries would remain leaking memory. This change fixes the memory leak by ensuring that only one dial status will only ever exist for each channel. The behavior during the scenario where multiple events are received has also been improved. For failure cases the first failure will be the dial status. If an answer dial status is received, though, it will take priority and the dial status for the channel will be answer. Memory usage has also been decreased by storing the minimal amount of information and the code has been cleaned up slightly. ASTERISK-25262 #close Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe	10 years ago
Mark Michelson	1fd3a7849e	ARI: Ensure proper channel state on operations. ARI was recently outfitted with operations to create and dial channels. This leads to the ability to try funny stuff. You could create a channel and then immediately try to play back media on it. You could create a channel, dial it, and while it is ringing attempt to make it continue in the dialplan. This commit attempts to fix this by adding a channel state check to operations that should not be able to operate on outbound channels that have not yet answered. If a channel is in an invalid state, we will send a 412 response. ASTERISK-26047 #close Reported by Mark Michelson Change-Id: I2ca51bf9ef2b44a1dc5a73f2d2de35c62c37dfd8	10 years ago
Mark Michelson	10019dc70c	test_http_media_cache: Fix failing test. The retrieve_cache_control_directives test has been failing occasionally in Jenkins. The apparent failure occurs when attempting to validate the expiration of the retrieved file. After reproducing, the problem was pretty clear. At the beginning of the test, the current time is retrieved. The seconds value of this timestamp is X. When the file is retrieved, res_http_media_cache calculates the expiration and in doing so retrieves the current time. In most cases, since the test executes quickly, it will also retrieve a timestamp with X seconds. However, if the test starts very near to when the timestamp seconds are set to increment, res_http_media_cache may retrieve a timestamp with X+1 seconds instead. The test attempted to account for this by allowing a tolerance of 1 second when validating the expiration. However, the problem was that the comparisons being used in the validation used > and < operations. This meant that values that fell within the tolerance (because they equaled the upper bound of the tolerance) would fail. The solution is to use >= and <= operators in the expiration validation. However, I estimated that while the one second tolerance should be fine on most machines, it would still be possible on a very slow machine to end up falling outside the one second tolerance. So I have also relaxed the tolerance of expiration validation to be three seconds instead. The final change here is to add a debug message when validating expiration so that we can see what values are being compared. ASTERISK-25959 #close Reported by Joshua Colp Change-Id: Ic1a0e10722c1c5d276d5a4d6a67136d6ec26c247	10 years ago
Timo Teräs	56bdf048d2	Add support for OGG/Speex file format ASTERISK-18995 #close Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a	10 years ago
zuul	0388c40b8c	Merge "chan_pjsip: Lock channel when checking for RTP changes."	10 years ago
George Joseph	f0855358a6	cdr.c: Remove assert in base_process_dial_end Scenario: Caller blonde transfer Bob calls Charlie who answers. Bob puts Charlie on hold and calls Alice. Before Alice answers, Bob transfers Charlie to Alice. Charlie's channel triggers an assert because he gets an "ANSWERED" event even though he never dialed anything. With recent changes to dial events, this is now a valid scenario so the assert needed to be removed. ASTERISK-26103 #close Change-Id: I2679b517b696e7952ab7fb29403df9140e7d1de2	10 years ago
Mark Michelson	cdb7edbe7b	chan_pjsip: Lock channel when checking for RTP changes. bridge_native_rtp can call into an RTP-capable channel driver in order for the driver to update information about who the channel is communicating with. For SIP channel drivers, this means deactivating RTCP and sending a reinvite so that the endpoints can communicate directly. bridge_native_rtp does the right thing and has the channel locked when calling into the channel driver. chan_pjsip can't alter session properties in this thread, though. chan_pjsip queues a task on the session serializer in order to update properties there. The problem is that this queued task was not locking the channel. This meant that the queued task could attempt to deactivate RTCP at the same time that the channel thread was attempting to process an incoming RTCP packet. This could lead to a crash. This patch fixes the issue by locking the channel in the queued task when altering RTP properties. ASTERISK-26092 #close Reported by Niklas Larsson Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159	10 years ago
Richard Mudgett	04ec9c745e	res_pjsip_registrar.c: Eliminate rx REGISTER request race condition. This patch fixes a race condition processing received REGISTER requests and their retransmissions caused by REGISTER requests being processed by two threads. The "sip_transaction Unable to register REGISTER transaction (key exists)" message is a notable symptom of this issue. This issue was more likely to happen before the pjsip/distributor serializers were created. Instead of steps one and two below placing the REGISTER messages into the same pjsip/distributor they were placed in random pjsip/default serializers. 1) REGISTER requests come in and get placed on the pjsip/distributor serializer. 2) Before the first request is processed a retransmission comes in and is placed on the same pjsip/distributor serializer. 3) The first request goes up the pjsip stack and is then shunted off to the pjsip/aor/<aor> serializer. 4) Before the first request is completed processing in the pjsip/aor/<aor> serializer, the second request goes up the pjsip stack and is also shunted off to the pjsip/aor/<aor> serializer. 5) The first request completes processing and sends out its response. 6) The second request completes processing and tries to send out its response but pjlib complains that the REGISTER transaction key already exists. 7) Sadness ensues. * The race is eliminated by removing the pjsip/aor/<aor> serializer and continuing the processing in the pjsip/distributor serializer. Now any retransmissions queued in the pjsip/distributor serializer will be processed after the first message is completely processed. ASTERISK-26088 #close Reported by: Richard Mudgett Change-Id: I842d714346088bf717ea27437f1dd85bff0bab5a	10 years ago
Richard Mudgett	dcfef53ee2	stasis: Add setting subscription congestion levels. Stasis subscriptions and message routers create taskprocessors to process the event messages. API calls are needed to be able to set the congestion levels of these taskprocessors for selected subscriptions and message routers. * Updated CDR, CEL, and manager's stasis subscription congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty call setup/teardown activity from triggering the taskprocessor overload alert. CDRs in particular need an extra high congestion level because they can take awhile to process the stasis messages. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: Id0a716394b4eee746dd158acc63d703902450244	10 years ago
Richard Mudgett	4879cd875c	sorcery: Add setting object type congestion levels. Sorcery creates taskprocessors for object types to process object observer callbacks. An API call is needed to be able to set the congestion levels of these taskprocessors for selected object types. * Updated PJSIP's contact and contact_status sorcery object type observer default congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty register/unregister and subscribe/unsubscribe activity from triggering the taskprocessor overload alert. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6	10 years ago
Richard Mudgett	2cd67d5b07	taskprocessors: Implement high/low water mark alerts. When taskprocessors get backed up, there is a good chance that we are being overloaded and need to defer adding new work to the system. * Implemented a high/low water alert mechanism for modules to check if the system is being overloaded and take appropriate action. When a taskprocessor is created it has default congestion levels set. A taskprocessor can later have those congestion levels altered for specific needs if stress testing shows that the taskprocessor is a symptom of overloading or needs to handle bursty activity without triggering an overload alert. * Add CLI "core show taskprocessor" low/high water columns. * Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was never a good thing to use when creating a taskprocessor because of the nature of how its references needed to be cleaned up on a partial creation. * Made res_pjsip's distributor check if the taskprocessor overload alert is active before placing a message representing brand new work onto a distributor serializer. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I182f1be603529cd665958661c4c05ff9901825fa	10 years ago
Richard Mudgett	c966a035e0	res_pjsip_session: Use distributor serializer for incoming calls. We must continue using the serializer that the original INVITE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. Outgoing call legs create the pjsip/outsess/<endpoint> serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc	10 years ago
Richard Mudgett	5b7b16a87f	res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer. * Resolves potential reentrancy problems if system restarted in the middle of subscription message transactions. * Fixes memory leak recreating persistent subscriptions when the subscription resource tree could not be created. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be	10 years ago
Richard Mudgett	c2ae49249c	res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions. We must continue using the serializer that the original SUBSCRIBE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. The "sip_transaction Unable to register SUBSCRIBE transaction (key exists)" message is a notable symptom of this issue. Outgoing subscriptions still create the pjsip/pubsub/<endpoint> serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0	10 years ago
Richard Mudgett	2ff26e9746	pjsip_distributor.c: Consistently pick a serializer for messages. Incoming messages that are not part of a dialog or a recognized response to one of our requests need to be sent to a consistent serializer. Under load we may be queueing retransmissions before we can process the original message. We don't need to throw these messages onto random serializers and cause reentrancy and message sequencing problems. * Created a pool of pjsip/distributor serializers that get picked by hashing the call-id and remote tag strings of the received messages. * Made ast_sip_destroy_distributor() destroy items in the reverse order of creation. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I2ce769389fc060d9f379977f559026fbcb632407	10 years ago
Richard Mudgett	df2791da8f	pjsip_distributor.c: Ignore messages until fully booted. We should not be processing any incoming messages until we are fully booted. We may not have dialplan or other needed configuration loaded yet. ASTERISK-26089 #close Reported by: Scott Griepentrog ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264	10 years ago
George Joseph	d21a77b325	build: Fix ast_sockaddr initialization to be more portable A change to glibc 2.22 changed the order of the sockadddr_storage members which caused the places where we do an initialization of ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those initializers (which we shouldn't have been using anyway) have been replaced with memsets. Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4	10 years ago
Joshua Colp	fbece11a0c	Merge "translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs."	10 years ago
Joshua Colp	2525563438	Merge "chan_sip: No rtpmap for static RTP payload IDs in SDP."	10 years ago
Joshua Colp	7eb3a3357c	Merge "BuildSystem: Avoid 'ar cru' and use 'ar cr' instead."	10 years ago
Joshua Colp	20be856b51	Merge "Detect and use proper libraries for musl toolchains"	10 years ago
Joshua Colp	5c949d009e	Merge "Fixes to include signal.h"	10 years ago
Joshua Colp	216f78c0ce	Merge "Make use of GLOB_BRACE and GLOB_NOMAGIC optional"	10 years ago
Joshua Colp	6ef3094239	Merge "res_hep_{pjsip\|rtcp}: Decline module loads if res_hep had not loaded"	10 years ago
Joshua Colp	1ead09dcb1	Merge "Fix res_search usage"	10 years ago
Joshua Colp	7bcccd4db3	Merge "Fix #include poll.h and sys/cdefs.h"	10 years ago
Timo Teräs	72d190eb69	Detect and use proper libraries for musl toolchains Change-Id: I8d9b212f70813404b82918a3f99439e500d4bfcb	10 years ago
Timo Teräs	39b69ab537	Fixes to include signal.h POSIX defines signal.h. sys/signal.h should not be used as it is c-library internal header which may or may not exist. Notably with musl it generates warning of being incorrect. Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc	10 years ago
Matt Jordan	7f5ca67e5f	res_hep_{pjsip\|rtcp}: Decline module loads if res_hep had not loaded A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not loaded and does not have a configuration file. Previously when this occurred, checks were put in to see if the configuration was loaded successfully. While this is a good idea - and has been added to the offending function in res_hep - the reality is res_hep_pjsip and res_hep_rtcp have no business running if res_hep isn't also running. As such, this patch also adds a function to res_hep that returns whether or not it successfully loaded. Oddly enough, ast_module_check returns "everything is peachy" even if a module declined its load - so it cannot be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this function to see if they should continue to load; if it fails, they decline their load as well. ASTERISK-26096 #close Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea	10 years ago
Joshua Colp	5164e1fd85	Merge "chan_rtp.c: Simplify options to UnicastRTP channel creation."	10 years ago
Joshua Colp	71f70f5a07	Merge "apps/app_voicemail.c and main/say.c: Add support for Icelandic language"	10 years ago
Joshua Colp	643dd462ee	Merge "ari/resource_channels: Add 'formats' to channel create/originate"	10 years ago
Alexander Traud	784c18128b	chan_sip: No rtpmap for static RTP payload IDs in SDP. This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over UDP, if many codecs are allowed in Asterisk. This new feature is enabled together with the optional feature compactheaders=yes via the file sip.conf. ASTERISK-25578 #close Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044	10 years ago
Joshua Colp	33787459c3	Merge "res_odbc: Implement a connection pool."	10 years ago
Joshua Colp	31a5c28339	res_odbc: Implement a connection pool. Testing has shown that our usage of UnixODBC is problematic due to bugs within UnixODBC itself as well as the heavy weight cost of connecting and disconnecting database connections, even when pooling is enabled. For users of UnixODBC 2.3.1 and earlier crashes would occur due to insufficient protection of the disconnect operation. This was fixed in UnixODBC 2.3.2 and above. For users of UnixODBC 2.3.3 and higher a slow-down would occur under heavy database use due to repeated connection establishment. A regression is present where on each connection the database configuration is cached again, with the cache growing out of control. The connection pool implementation present in this change helps to mitigate these issues by reducing how much we connect and disconnect database connections. We also solve the issue of crashes under UnixODBC 2.3.1 by defaulting the maximum number of connections to 1, returning us to the previous working behavior. For users who may have a fixed version the maximum concurrent connection limit can be increased helping with performance. The connection pool works by keeping a list of active connections. If the connection limit has not been reached a new connection is established. If the connection limit has been reached then the request waits until a connection becomes available before continuing. ASTERISK-26074 #close ASTERISK-26054 #close Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff	10 years ago
Vasil Kolev	80ff7912a1	chan_sip: bigger buffers for headers, better failure mode Currently chan_sip can give weird messages if the contacts don't fit in the From: or To: headers. This fix changes the from,to and invite variables to use ast_str, allocates and deallocates them and resizes them if needed. ASTERISK-26069 #close Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3	10 years ago
Örn Arnarson	60caebc738	apps/app_voicemail.c and main/say.c: Add support for Icelandic language Icelandic has some weird grammar rules when dealing with dates and numbers. There are different genders used depending on which number you're dealing with, and only a handful of numbers do change depending on the gender. There is also an implied gender in several cases. This patch was originally written for asterisk 1.6, and has been in use for several years without crashes. I cleaned it up a bit and rewrote what was necessary for Asterisk 13. The functions were copied from other similar languages and modified where appropriate. If i recall correctly, the German and Danish functions were used as a base. ASTERISK-26087 Reported by: Örn Arnarson Tested by: Örn Arnarson Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665	10 years ago
Alexander Traud	52120204c9	res_srtp: Instead of libSRTP use OpenSSL as random source. Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore. Therefore, the symbol RAND_bytes is used instead of crypto_get_random. ASTERISK-24436 #close Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96	10 years ago
Alexander Traud	da943ec5c0	BuildSystem: Avoid 'ar cru' and use 'ar cr' instead. In several internal library projects, the files are archived with the help of 'ar cr'. Only the projects editline and the Objective Open H.323 stack implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier ignored since `D' is the default (see `U')". For consistency and to avoid this message all projects use 'ar cr' now. ASTERISK-26091 #close Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40	10 years ago
Richard Mudgett	dca052e531	chan_rtp.c: Simplify options to UnicastRTP channel creation. Change the awkward and not as flexible UnicastRTP options format From: Dial(UnicastRTP/127.0.0.1[/[<engine>][/[<codec>]]]) To: Dial(UnicastRTP/127.0.0.1[/[<options>]]) Where <options> can be standard Asterisk flag options: c(<codec>) - Specify which codec/format to use such as 'ulaw'. e(<engine>) - Specify which RTP engine to use such as 'asterisk'. More option flags can be easily added later such as the codec's RTP payload type to use when the codec does not have a static payload type defined. Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9	10 years ago
Jaco Kroon	5bfef2a8b4	translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs. ASTERISK-25629 #close Change-Id: Ibfcf0670e094e9718d82fd9920f1fb2dae122006	10 years ago
Alexei Gradinari	3e8d523d88	core/dial: New channel variable FORWARDERNAME Added a new channel variable FORWARDERNAME which indicates which channel was responsible for a forwarding requests received on dial attempt. Fixed a bug in the app_queue: FORWARD_CONTEXT is not used. ASTERISK-26059 #close Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2	10 years ago
George Joseph	a2f820e8dc	ari/resource_channels: Add 'formats' to channel create/originate If you create a local channel and don't specify an originator channel to take capabilities from, we automatically add all audio formats to the new channel's capabilities. When we try to make the channel compatible with another, the "best format" functions pick the best format available, which in this case will be slin192. While this is great for preserving quality, it's the worst for performance and overkill for the vast majority of applications. In the absense of any other information, adding all formats is the correct thing to do and it's not always possible to supply an originator so a new parameter 'formats' has been added to the channel create/originate functions. It's just a comma separated list of formats to make availalble for the channel. Example: "ulaw,slin,slin16". 'formats' and 'originator' are mutually exclusive. To facilitate determination of format names, the format name has been added to "core show codecs". ASTERISK-26070 #close Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b	10 years ago

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