asterisk

Commit Graph

Author	SHA1	Message	Date
Alexander Traud	f10fc135d4	res_pjsip_endpoint_identifier_ip: Unregister the module for headers. Asterisk uses Reference Counting to track whether a module can be unloaded. Every consumer who requires a module, increases the reference count. When the consumer goes, is unloaded itself, it has to decrease the reference count on all its used/required modules. That way core stop gracefully works on the command-line interface (CLI): One module after the other is unloaded. A recent change broke this for the module res_pjsip. ASTERISK-27861 Change-Id: I261abcb411d026bbb0691cc78f28300bfd3103a3	8 years ago
Joshua Colp	60ce5d0003	Merge "cli: Display correct unit for HTTP timeout in "manager show settings"."	8 years ago
Joshua Colp	195af35026	Merge "Fix GCC 8 build issues."	8 years ago
Joshua Colp	1cc4313d32	Merge "rtp_engine: Remove the double assigned RTP payload ID of H.263+."	8 years ago
Joshua Colp	c0c79997f8	Merge "git: Ignore *.orig."	8 years ago
Joshua Colp	bc4b3c535d	Merge "rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code."	8 years ago
Joshua Colp	a103221de2	Merge "pjsip: Rewrite OPTIONS support with new eyes."	8 years ago
Alexander Traud	71d1e8d8c8	rtp_engine: Remove the double assigned RTP payload ID of H.263+. Mantis-3709 (Commit `68ff3c3`, Asterisk 1.2) added support for the video format H.263+. For this, the RTP payload ID 103 got assigned statically. Commit `f1aadc8` assigned another payload ID 98 for this format in Asterisk 1.6. Change-Id: I90e35b158487f8f1f8187da6241b54cd3b74e667	8 years ago
Corey Farrell	4722a653f4	cli: Display correct unit for HTTP timeout in "manager show settings". HTTP timeout is in seconds, not minutes. ASTERISK-27852 #close Change-Id: Ie6640835cb07307555741f9b559c2eb876d9343e	8 years ago
Alexander Traud	263637a38d	rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code. Change-Id: Ica089d4507a27ddfc4ce3a88d697ffbef378de48	8 years ago
Corey Farrell	b5914d90ac	Fix GCC 8 build issues. This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	8 years ago
Alexander Traud	919b0eb3f2	rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. This issue affected only installations with rtp_use_dynamic=yes in asterisk.conf which is the default since Asterisk 15. Codec 2 and SiLK were built-in examples of media formats which were affected. ASTERISK-27850 Reported by: Dinis Brazão, Selene Feigl Change-Id: I08c1e76433a67e4350141d38cacf3a1cb5086496	8 years ago
Joshua Colp	6773ea9e39	Merge "makeopts.in: Remove unused/undefined AST_MARCH_NATIVE."	8 years ago
Joshua Colp	5437b3932d	Merge "sip_to_pjsip: Enable python3 compatibility."	8 years ago
Joshua Colp	1351f42363	Merge "res_hep: Adds hostname resolution support for capture_address"	8 years ago
Jenkins2	179c794879	Merge "app_macro: Prevent infinite loop in find_matching_priority."	8 years ago
Corey Farrell	2e37684913	git: Ignore *.orig. This prevents accidental commit of files created by patch. Change-Id: I68380db61f0f9d620046f719ccd978811d0e9964	8 years ago
Alexander Traud	2d81709ab1	sip_to_pjsip: Enable python3 compatibility. The script remains compatible with Python 2.7 but now also works with Python 3.3 and newer; to ease the migration from chan_sip to chan_pjsip. ASTERISK-27811 Change-Id: I59cc6b52a1a89777eebcf25b3023bdf93babf835	8 years ago
Corey Farrell	cea87fe7b8	makeopts.in: Remove unused/undefined AST_MARCH_NATIVE. Change-Id: I617a96ebb83ec99f5d3176bbbee2d2a272ccb203	8 years ago
Jaco Kroon	9f1e1d153a	manager: fix digest auth for ami/http mechanism. Due to a fixed size buffer the digest authentication could be incorrectly calculated if a large URI was provided, causing authentication failure. The buffer is now dynamically allocated to allow any size URI within the normal limits of the HTTP request size. ASTERISK-27841 Change-Id: I660609db13b8f9e5f9567f339dd804f4985d41b3	8 years ago
Jenkins2	d83a37f0cc	Merge "stream: Make the topology a reference counted object."	8 years ago
Corey Farrell	d855658f23	app_macro: Prevent infinite loop in find_matching_priority. Use AST_PBX_MAX_STACK to escape if we recurse 128 times. This will prevent crash if dialplan contains an include loop. Log an error when this occurs, at most one message per call to Macro() so we avoid logger spam. ASTERISK-26570 #close Change-Id: I6c71b76998c31434391b150de055ae9a531e31da	8 years ago
Tzafrir Cohen	f4c360143b	cdr_mysql: my_connect_db(): reduce indentation ASTERISK-27572 Change-Id: I00bd5363ac94c764c56d8626a5945ed7f3934fcb	8 years ago
Tzafrir Cohen	2e44adf1c3	cdr_mysql: split mysql init out of my_load_module Split out mysql connection parts to a separate my_connect_db(). ASTERISK-27572 Change-Id: If2ee676056067cc693ff08be68ee4944bf35b49f	8 years ago
Matthew Fredrickson	8f55f7c333	res_hep: Adds hostname resolution support for capture_address Previously, only an IP address would be accepted for the capture_address config setting in hep.conf. This change allows capture_address to be a resolvable hostname or an IP address. ASTERISK-27796 #close Reported-By: Sebastian Gutierrez Change-Id: I33e1a37a8b86e20505dadeda760b861a9ef51f6f	8 years ago
Jenkins2	57ad488451	Merge "res_ari: Remove requirement that body exists when debug is on."	8 years ago
Jenkins2	dcaaae6cd1	Merge "iostreams: Add some documentation for the ast_iostream_* functions"	8 years ago
Joshua Colp	11f5aba43b	Merge "chan_dahdi: Configurable dialed digit timeouts"	8 years ago
Jenkins2	85f894a8e0	Merge "pbx_lua: Support displaying lua error message if no debug table exists"	8 years ago
Jenkins2	8e228fc138	Merge "res_pjsip/pjsip_distributor.c: Add missing off-nominal request response."	8 years ago
Joshua Colp	7528b86cad	stream: Make the topology a reference counted object. The stream topology has no lock of its own resulting in another lock protecting it in some way (for example the channel lock). If multiple channels are being juggled at the same time this can be problematic. This change makes the topology a reference counted object instead which guarantees it will remain valid even without the channel lock being held. Change-Id: I4f4d3dd856a033ed55fe218c3a4fab364afedb03	8 years ago
Jenkins2	0565a6c909	Merge "Add the ability to read the media file type from HTTP header for playback"	8 years ago
Tzafrir Cohen	6301531416	chan_dahdi: Configurable dialed digit timeouts Analog phones dial overlap dialing and it is chan_dahdi's job to read the numbers. It has three timeout constants that this commit converts to channel-level configuration options: * firstdigit_timeout: Default time (ms) to detect first digit * interdigit_timeout: Default time (ms) to detect following digits * matchdigit_timeout: Default time (ms) to wait in case of ambiguous match. This happens when the dialed digits match a number in the current context but are also the prefix of another number. Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213	8 years ago
Jenkins2	e538fc8e86	Merge "res_rtp_asterisk: Always update SRTP on local SSRC change."	8 years ago
Jenkins2	706b899358	Merge "pjsip: Increase maximum number of usable ciphers & other cleanups"	8 years ago
Joshua Colp	de3ca9bada	res_ari: Remove requirement that body exists when debug is on. The "ari set debug" code for incoming requests incorrectly assumed that all requests would contain a body. If one did not exist the request would be incorrectly rejected. The response that was sent was also incomplete as an incorrect function was used to construct the response. The code has now been changed to no longer require a request to have a body and the response updated to use the correct function. ASTERISK-27801 Change-Id: I4eef036ad54550a4368118cc348765ecac25e0f8	8 years ago
Sean Bright	069a0b7593	iostreams: Add some documentation for the ast_iostream_* functions Change-Id: Id71b87637f0a484eb5a1cd26c3d1c7c15c7dcf26	8 years ago
Sean Bright	239074c759	pjsip: Increase maximum number of usable ciphers & other cleanups * Increase maximum number of ciphers from 100 to 256 (or whatever PJ_SSL_SOCK_MAX_CIPHERS is #define'd to) * Simplify logic in cipher_name_to_id() * Make signed/unsigned comparison consistent Re: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=897412 Reported by: Ondřej Holas Change-Id: Iea620f03915a1b873e79743154255c3148a514e7	8 years ago
Richard Mudgett	11b7de82c5	res_pjsip/pjsip_distributor.c: Add missing off-nominal request response. Change-Id: I389579b39c523d1d1e8ce020ef549a8bb5781c9b	8 years ago
Richard Mudgett	6cab3c836a	res_pjsip/pjsip_distributor.c: Pull some assignments out of if tests. Change-Id: I3d30d638b53a4bbe9bf9aad853c649d583894112	8 years ago
Joshua Colp	afdca5c68c	res_rtp_asterisk: Always update SRTP on local SSRC change. When the local SSRC changes we need to update the SRTP information so that the proper key is used. This is commonly done as a result of bridging two channels together. Previously we only updated the SRTP information if media had already flowed, but in practice the channel driver may have already performed SRTP negotiation and set up the previous SSRC. We now always do it on a local SSRC change. ASTERISK-27795 ASTERISK-27800 Change-Id: Ia7c8e74c28841388b5244ac0b8fd6c1dc6ee4c10	8 years ago
Gaurav Khurana	0827d5cc53	Add the ability to read the media file type from HTTP header for playback How it works today: media_cache tries to parse out the extension of the media file to be played from the URI provided to Asterisk while caching the file. What's expected: Better will be to have Asterisk get extension from other ways too. One of the common ways is to get the type of content from the CONTENT-TYPE header in the HTTP response for fetching the media file using the URI provided. Steps to Reproduce: Provide a URL of the form: http://host/media/1234 to Asterisk for media playback. It fails to play and logs show the following error line: [Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c: File http://host/media/1234 does not exist in any format Scenario this issue is blocking: In the case where the media files are stored in some cloud object store, following can block the media being played via Asterisk: Cloud storage generally needs authenticated access to the storage. The way to do that is by using signed URIs. With the signed URIs there's no way to preserve the name of the file. In most cases Cloud storage returns a key to access the object and preserving file name is also not a thing there ASTERISK-27286 Reporter: Gaurav Khurana Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89	8 years ago
Jenkins2	f633af89c1	Merge "bridge_softmix: Fix sporadic incorrect video stream mapping."	8 years ago
George Joseph	3bad41257b	Merge "BuildSystem: Add DragonFly BSD."	8 years ago
Jenkins2	8e368d0eaf	Merge "translate: generic plc not filled in after translation"	8 years ago
George Joseph	5dd6fe478c	Merge "app_sendtext: Enhance SendText to support Enhanced Messaging"	8 years ago
Christof Lauber	9c9f314f64	pbx_lua: Support displaying lua error message if no debug table exists The lua_error_function assumed that lua's debug table and traceback function are always accessible, which is not the case. This fixes the error message 'Error in the lua error handler' triggred by switch exec() function. If this happens lua's error message is shown without traceback. Change-Id: I34ba0a098f1ae06a3af7b4d1b098bd43f42f96c8	8 years ago
Joshua Colp	882e79b77e	pjsip: Rewrite OPTIONS support with new eyes. The OPTIONS support in PJSIP has organically grown, like many things in Asterisk. It has been tweaked, changed, and adapted based on situations run into. Unfortunately this has taken its toll. Configuration file based objects have poor performance and even dynamic ones aren't that great. This change scraps the existing code and starts fresh with new eyes. It leverages all of the APIs made available such as sorcery observers and serializers to provide a better implementation. 1. The state of contacts, AORs, and endpoints relevant to the qualify process is maintained. This state can be updated by external forces (such as a device registering/unregistering) and also the reload process. This state also includes the association between endpoints and AORs. 2. AORs are scheduled and not contacts. This reduces the amount of work spent juggling scheduled items. 3. Manipulation of which AORs are being qualified and the endpoint states all occur within a serializer to reduce the conflict that can occur with multiple threads attempting to modify things. 4. Operations regarding an AOR use a serializer specific to that AOR. 5. AORs and endpoint state act as state compositors. They take input from lower level objects (contacts feed AORs, AORs feed endpoint state) and determine if a sufficient enough change has occurred to be fed further up the chain. 6. Realtime is supported by using observers to know when a contact has been registered. If state does not exist for the associated AOR then it is retrieved and becomes active as appropriate. The end result of all of this is best shown with a configuration file of 3000 endpoints each with an AOR that has a static contact. In the old code it would take over a minute to load and use all 8 of my cores. This new code takes 2-3 seconds and barely touches the CPU even while dealing with all of the OPTIONS requests. ASTERISK-26806 Change-Id: I6a5ebbfca9001dfe933eaeac4d3babd8d2e6f082	8 years ago
Jenkins2	9c430569d4	Merge "bridge_softmix: Forward TEXT frames"	8 years ago
Joshua Colp	2cef65dc77	Merge "BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD."	8 years ago

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