asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	c1e49720fa	test.c: Make output jUnit compatible Separate "name" into "classname" and "name". Use '.' for classname separator instead of '/'. Prefix reserved words with '_'. Wrap output with a top-level "testsuites" element. Change-Id: Iec1a985eba1c478e5c1d65d5dfd95cb708442099	8 years ago
Joshua Colp	454aa952fa	Merge "main/cdr.c: Alleviate CDR deadlock"	8 years ago
Matthew Fredrickson	db02218db2	main/cdr.c: Alleviate CDR deadlock There is a rare case (do to the infrequent timing involved) where CDR submission threads in batch mode can deadlock with a currently running CDR batch process. This patch should remove the need for holding the lock in the scheduler and should clean a few code paths up that inconsistently submitted new work to the CDR batch processor. ASTERISK-27909 Change-Id: I6333e865db7c593c102c2fd948cecdb96481974d Reported-by: Denis Lebedev	8 years ago
Richard Mudgett	7a238fe74d	AMI SendText action: Fix to use correct thread to send the text. The AMI action was directly sending the text to the channel driver. However, this makes two threads attempt to handle media and runs afowl of CHECK_BLOCKING. * Queue a read action to make the channel's media handling thread actually send the text message. This changes the AMI actions success/fail response to just mean the text was queued to be sent not that the text actually got sent. The channel driver may not even support sending text messages. ASTERISK-27943 Change-Id: I9dce343d8fa634ba5a416a1326d8a6340f98c379	8 years ago
George Joseph	e3585353f6	res_pjsip_messaging: Allow application/* for in-dialog MESSAGEs In addition to text/* content types, incoming_in_dialog_request now accepts application/* content types. Also fixed a length issue when copying the body text. It was one character short. ASTERISK-27942 Change-Id: I4e54d8cc6158dc47eb8fdd6ba0108c6fd53f2818	8 years ago
Alexander Traud	675e2ddb49	uuid: Enable UUID in Solaris 11. ASTERISK-27933 Reported by: bautsche Change-Id: I9b8362824efbfb2a16981e46e85f7c8322908c49	8 years ago
Jenkins2	bbd962bc4b	Merge "utils: Avoid an unused variable in Solaris 11."	8 years ago
George Joseph	d87631d21f	Merge changes from topic 'ASTERISK-27625' * changes: channel.c: Make CHECK_BLOCKING() save thread LWP id for messages. channel.c: Fix usage of CHECK_BLOCKING() autoservice: Don't start channel autoservice if the thread is a user interface.	8 years ago
George Joseph	98bd5908a2	Merge "ARI POST DTMF: Make not compete with channel's media thread."	8 years ago
George Joseph	46c1f81fad	Merge "AMI PlayDTMF Action: Make not compete with channel's media thread."	8 years ago
Alexander Traud	a5c53bd323	utils: Avoid an unused variable in Solaris 11. With ./configure --enable-dev-mode[=noisy], the build fails because every warning gets an error. Therefore, Asterisk has to be free of warnings and this variable must go. Change-Id: I63dd2bc4833b9bdb04602f83422d16caf289d46a	8 years ago
Jenkins2	5065cb2c24	Merge "Fix some doxygen and curly placement."	8 years ago
Jenkins2	8892a9b8d9	Merge "Dialplan functions: Fix some channel autoservice misuse."	8 years ago
Richard Mudgett	eb8bbe660e	channel.c: Make CHECK_BLOCKING() save thread LWP id for messages. * Removed an unnecessary call to ast_channel_blocker_set() in __ast_read(). ASTERISK-27625 Change-Id: I342168b999984666fb869cd519fe779583a73834	8 years ago
Richard Mudgett	da54605b8a	ARI POST DTMF: Make not compete with channel's media thread. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e	8 years ago
Richard Mudgett	7d874c1af7	AMI PlayDTMF Action: Make not compete with channel's media thread. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905	8 years ago
Richard Mudgett	080508d2eb	channel.c: Fix usage of CHECK_BLOCKING() The CHECK_BLOCKING() macro is used to indicate if a channel's handling thread is about to do a blocking operation (poll, read, or write) of media. A few operations such as ast_queue_frame(), soft hangup, and masquerades use the indication to wake up the blocked thread to reevaluate what is going on. ASTERISK-27625 Change-Id: I4dfc33e01e60627d962efa29d0a4244cf151a84d	8 years ago
Richard Mudgett	0989b63047	autoservice: Don't start channel autoservice if the thread is a user interface. Executing dialplan functions from either AMI or ARI by getting a variable could place the channel into autoservice. However, these user interface threads do not handle the channel's media so we wind up with two threads attempting to handle the media. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: If2dc94ce15ddabf923ed1e2a65ea0ef56e013e49	8 years ago
Richard Mudgett	91c3ac19cb	Dialplan functions: Fix some channel autoservice misuse. * Fix off nominal paths leaving the channel in autoservice. * Remove unnecessary start/stop channel autoservice. * Fix channel locking around a channel datastore search. Change-Id: I7ff2e42388064fe3149034ecae57604040b8b540	8 years ago
Richard Mudgett	720c2d1da2	Fix some doxygen and curly placement. Change-Id: I9a784a7c804120a8fa826c2a4cb9957e4b0b2fc8	8 years ago
Richard Mudgett	a470bb9e27	channel: Fix some more unprotected channel flag setting. Change-Id: I34c3b1201b1de539945bcfdcb264fff30332d48c	8 years ago
Joshua Colp	41175caee0	rtp: Don't negotiate dynamic codecs using payload. In Asterisk there are some dynamic codecs that have a fixed payload number. This number was being improperly used to negotiate the codec, instead of using the name and sample rate. This could result in the wrong payload number being negotiated for a codec. This change makes it so that only static payloads will be negotiated using their payload number. ASTERISK-27848 Change-Id: Ia865830170fd3f808cdb33104f3d4c4ffdc77570	8 years ago
Sean Bright	b649682caa	AST-2018-007: iostreams potential DoS when client connection closed prematurely Before Asterisk sends an HTTP response (at least in the case of errors), it attempts to read & discard the content of the request. If the client lies about the Content-Length, or the connection is closed from the client side before "Content-Length" bytes are sent, the request handling thread will busy loop. ASTERISK-27807 Change-Id: I945c5fc888ed92be625b8c35039fc6d2aa89c762	8 years ago
Joshua Colp	c98c1b3f74	Merge "bridge_channel.c: Fix Deadlock when using Local channels and fax gateway"	8 years ago
Joshua Colp	2151903a16	Merge "tcptls: Allow OpenSSL configured with no-dh."	8 years ago
George Joseph	76339b1962	Merge "tcptls.h: Repair ./configure --with-ssl=PATH."	8 years ago
George Joseph	99aad2f0af	Merge "tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated."	8 years ago
Pirmin Walthert	e078558038	bridge_channel.c: Fix Deadlock when using Local channels and fax gateway ast_indicate is invoked with the bridge locked. As ast_indicate locks the other end of the bridge as well this can lead to a deadlock in some situations. (Especially when a different thread does the same in the reverse order). This patch calls ast_indicate after unlocking the bridge which fixes the deadlock. Calling ast_indicate with these parameters without locking the bridge should be safe as this is done at different places without a bridge lock. ASTERISK-27094 #close Reported-by: David Brillert Change-Id: I5f86c1e2ce75b9929a36ab589b18c450e62ea35f	8 years ago
George Joseph	437ab41881	app_sendtext: Allow content types other than text/plain There was no real reason to limit the conteny type to text/plain other than that's what it was limited to before. Now any text/* content type will be allowed for channel drivers that don't support enhanced messaging and any type will be allowed for channel drivers that do support enhanced messaging. Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9	8 years ago
Joshua Colp	c63cd006ba	Merge "libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated."	8 years ago
Alexander Traud	24503fb600	tcptls.h: Repair ./configure --with-ssl=PATH. asterisk/tcptls.h was included (explicitly, implicitly, or transitively). Those inclusions got replaced by forward declarations. As side effect, the inclusions got completed. ASTERISK-27878 Change-Id: I9d102728e30336d6522e5e4ae9e964013a0835f7	8 years ago
Alexander Traud	d36338ce2b	tcptls: Allow OpenSSL configured with no-dh. Additionally, this change allows auto-negotiation of the elliptic curve/group for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect. ASTERISK-27876 Change-Id: I62c2aba4a630aefc231b71f646207e8c027d9497	8 years ago
Alexander Traud	91616f4524	tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. ASTERISK-27874 Change-Id: Ica65113511c7a1c13f7988e7d9e7d9e7f3f620dd	8 years ago
Joshua Colp	4ea98e49f1	Merge "rtp: Add support for RTP extension negotiation and abs-send-time."	8 years ago
Joshua Colp	fbb33ba6e8	Merge "tcptls: Repair ./configure --with-ssl=PATH."	8 years ago
Joshua Colp	ca9120a1f0	Merge "config.c: Fix successful DELETE treated as failure"	8 years ago
Joshua Colp	7e655b26d1	Merge "channel.c: Fix off nominal channel allocation failure path."	8 years ago
Joshua Colp	25764691b0	Merge "netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API"	8 years ago
Joshua Colp	a507c73a78	rtp: Add support for RTP extension negotiation and abs-send-time. When RTP was originally created it had the ability to place a single extension in an RTP packet. In practice people wanted to potentially put multiple extensions in one and so RFC 5285 (obsoleted by RFC 8285) came into existence. This allows RTP extensions to be negotiated with a unique identifier to be used in the RTP packet, allowing multiple extensions to be present in the packet. This change extends the RTP engine API to add support for this. A user of it can enable extensions and the API provides the ability to retrieve the information (to construct SDP for example) and to provide negotiated information (from SDP). The end result is that the RTP engine can then query to see if the extension has been negotiated and what unique identifier is to be used. It is then up to the RTP engine implementation to construct the packet appropriately. The first extension to use this support is abs-send-time which is defined in the REMB draft[1] and is a second timestamp placed in an RTP packet which is for when the packet has left the sending system. It is used to more accurately determine the available bandwidth. ASTERISK-27831 [1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 Change-Id: I508deac557867b1e27fc7339be890c8018171588	8 years ago
Richard Mudgett	1bec0c73b3	channel.c: Fix off nominal channel allocation failure path. __ast_channel_alloc_ap() had a failure exit path that hadn't setup the fd descriptors to -1 yet. The destructor would then attempt to close these fd's that had never been opened. Change-Id: Icf21093f36c60781e8cf6ee9d586536302af33e3	8 years ago
Alexei Gradinari	39632c7e00	config.c: Fix successful DELETE treated as failure The config engine destroy_func callback function returns the number of rows deleted or -1 on error. But the function ast_destroy_realtime_fields treated non-zero return values as error. ASTERISK-27863 Change-Id: Ied02b38e8196cb03043e609a0679feebd288d17b	8 years ago
Matthew Fredrickson	9f9dce05b2	netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API This function originally was used in chan_sip to enable some simplifying assumptions and eventually was copy and pasted into res_pjsip_logger and res_hep. Since it's replicated in three places, it's probably best to move it into the public netsock2 API for these modules to use. Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04	8 years ago
Alexander Traud	1424f42d25	libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated. Use CRYPTO_set_id_callback(.) only with OpenSSL 0.9.8 and older. ASTERISK-27867 Change-Id: Iadd58d5bf6f538eb224203970a4e88e26f259655	8 years ago
Alexander Traud	2228ae3f27	tcptls: Repair ./configure --with-ssl=PATH. SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 got discovered without honoring a PATH. ASTERISK-27865 Change-Id: I8cd358eed7411726d08fa7b01691bef122fbeb71	8 years ago
Kevin Harwell	357654313f	Merge "rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again."	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
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
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

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6236 Commits (fcc0a6fe8a25e758e2140c0fdb196cc1e2aa4e90)