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Author	SHA1	Message	Date
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	9 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	9 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	9 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	9 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	9 years ago
Joshua Colp	fbece11a0c	Merge "translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs."	9 years ago
Joshua Colp	2525563438	Merge "chan_sip: No rtpmap for static RTP payload IDs in SDP."	9 years ago
Joshua Colp	7eb3a3357c	Merge "BuildSystem: Avoid 'ar cru' and use 'ar cr' instead."	9 years ago
Joshua Colp	20be856b51	Merge "Detect and use proper libraries for musl toolchains"	9 years ago
Joshua Colp	5c949d009e	Merge "Fixes to include signal.h"	9 years ago
Joshua Colp	216f78c0ce	Merge "Make use of GLOB_BRACE and GLOB_NOMAGIC optional"	9 years ago
Joshua Colp	6ef3094239	Merge "res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded"	9 years ago
Joshua Colp	1ead09dcb1	Merge "Fix res_search usage"	9 years ago
Joshua Colp	7bcccd4db3	Merge "Fix #include poll.h and sys/cdefs.h"	9 years ago
Timo Teräs	72d190eb69	Detect and use proper libraries for musl toolchains ... Change-Id: I8d9b212f70813404b82918a3f99439e500d4bfcb	9 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	9 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	9 years ago
Joshua Colp	5164e1fd85	Merge "chan_rtp.c: Simplify options to UnicastRTP channel creation."	9 years ago
Joshua Colp	71f70f5a07	Merge "apps/app_voicemail.c and main/say.c: Add support for Icelandic language"	9 years ago
Joshua Colp	643dd462ee	Merge "ari/resource_channels: Add 'formats' to channel create/originate"	9 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	9 years ago
Joshua Colp	33787459c3	Merge "res_odbc: Implement a connection pool."	9 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	9 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	9 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	9 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	9 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	9 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	9 years ago
Jaco Kroon	5bfef2a8b4	translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs. ... ASTERISK-25629 #close Change-Id: Ibfcf0670e094e9718d82fd9920f1fb2dae122006	9 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	9 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	9 years ago
Joshua Colp	f7ce0f1832	Merge "core/manager: Add uptime field to FullyBooted"	9 years ago
Timo Teräs	538c6415c6	chan_sip: Support auth username for callbackextension feature ... ASTERISK-20527 #close Change-Id: I659cf7f00836a09d09d146ad226a40477d731239	9 years ago
Timo Teräs	797695c5cc	Make use of GLOB_BRACE and GLOB_NOMAGIC optional ... These flags are non-portable GNU extensions. Make their use optional. This fixes complication error on e.g. musl c-library based systems. Change-Id: I0aa06efc62aa8995f091445c8b762a75a91042f3	9 years ago
Timo Teräs	3c1fec8099	Fix res_search usage ... Resolver state is not part of res_search API. This fixes compilation error: dns.c:261:8: error: too many arguments to function 'res_search' ret = res_search(&dns_state, Change-Id: Ia600a58557040df83f744da3dde23225293845a5	9 years ago
Timo Teräs	9c1d95e873	Fix #include poll.h and sys/cdefs.h ... POSIX defines poll.h, sys/poll.h should not be used at is c-library internal header which may or may not exist. Notable in musl it generates warning of being incorrect. And add explict include of sys/cdefs.h where needed. Change-Id: I142930df53fe7585a06b854b6faddc5301e024be	9 years ago
Niklas Larsson	8a5c2e736c	core/manager: Add uptime field to FullyBooted ... Add Uptime and LastReload to event FullyBooted. ASTERISK-26058 #close Reported by: Niklas Larsson Change-Id: I909b330801c0990d78df9b272ab0adc95aecb15e	9 years ago
Joshua Colp	4505a59dc9	alembic: Fix migration. ... The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting to use UniqueConstraint and failing. It was not imported and after importing it also continued to fail. I've changed the script to use the explicit name of the constraint instead. Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9	9 years ago
Joshua Colp	48843a107d	Merge "pjsip_distributor.c: Use correct rdata info access method (Part 2)."	9 years ago
Joshua Colp	e863bca856	Merge "logging,cdr,cel: Fix stringfield memory leak."	9 years ago
Joshua Colp	58aef11dc1	Merge "pjproject_bundled: Move to pjproject 2.5"	9 years ago
Richard Mudgett	40d19f2e55	logging,cdr,cel: Fix stringfield memory leak. ... The stringfields refactor to allow adding stringfields to the end of a structure (f6f4cf459f) exposed some incomplete cleanup code by some stringfield users. The most noticeable leaker is the logging system where there is a leak for every log message generated. ASTERISK-26078 #close Reported by: Etienne Lessard Patches: jira_asterisk_26078_v13.patch (license #5621) patch uploaded by Richard Mudgett Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782	9 years ago
Joshua Colp	608e0267e8	Merge "Expand the scope of Dial Events"	9 years ago
Richard Mudgett	aec7916595	pjsip_distributor.c: Use correct rdata info access method (Part 2). ... The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799	9 years ago
Joshua Colp	ad31e5bb1c	Merge "followme: allow disabling callee prompt"	9 years ago
zuul	84a93b0d67	Merge "ARI: Re-implement the ARI dial command, allowing for early bridging."	9 years ago
zuul	dffbb2d7c3	Merge "res_pjsip_mwi_body_generator: Re-order the body items"	9 years ago
Mark Michelson	205a31f86c	Expand the scope of Dial Events ... Dial events up to this point have come in two flavors * A Dial event with no status to indicate that dialing has begun * A Dial event with a status to indicate that dialing has ended With this change, Dial events have been expanded to also give intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS". This is especially useful for ARI dialing, as it gives the application writer the opportunity to place a channel into an early bridge when early media is detected. AMI handles these in-progress dial events by sending a new event called "DialState" that simply indicates that dial state has changed but has not ended. ARI never distinguished between DialBegin and DialEnd, so no change was made to the event itself. Another change here relates to dial forwards. A forward-related event was previously only sent when a channel was successfully able to forward a call to a new channel. With this set of changes, if forwarding is blocked, we send a Dial event with a forwarding destination but no forwarding channel, since we were prevented from creating one. This is again useful for ARI since application writers can now handle call forward attempts from within their own application. ASTERISK-25925 #close Reported by Mark Michelson Change-Id: I42cbec7730d84640a434d143a0d172a740995543	9 years ago
Joshua Colp	11ea121cc8	Merge "res_pjsip: add "via_addr", "via_port", "call_id" to contact"	9 years ago
zuul	853350b5ea	Merge "res_pjsip: Add clarifying documentation to PJSIP_HEADER help text"	9 years ago