asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	5e22118a91	Merge "lock: Improve performance of DEBUG_THREADS." into 15	8 years ago
Joshua Colp	3be86d3f55	Merge "configure.ac: Check for unbound version >= 1.5" into 15	8 years ago
Corey Farrell	1ffbec4672	lock: Improve performance of DEBUG_THREADS. Add a volatile flag to lock tracking structures so we only need to use the global lock when first initializing tracking. Additionally add support for DEBUG_THREADS_LOOSE_ABI. This is used by astobj2.c to eliminate storage for tracking fields when DEBUG_THREADS is not defined. Change-Id: Iabd650908901843e9fff47ef1c539f0e1b8cb13b	8 years ago
George Joseph	a0d5f45baa	Merge "jansson: Backport fixes to bundled, use json_vsprintf if available." into 15	8 years ago
George Joseph	ff8f46bf75	Merge "rtp_engine: rtcp_report_to_json can overflow the ssrc integer value" into 15	8 years ago
George Joseph	4c636f5377	configure.ac: Check for unbound version >= 1.5 In order to do this and provide good feedback, a new macro was created (AST_EXT_LIB_EXTRA_CHECK) which does the normal check and path setups for the library then compiles, links and runs a supplied code fragment to do the final determination. In this case, the final code fragment compares UNBOUND_VERSION_MAJOR and UNBOUND_VERSION_MINOR to determine if they're greater than or equal to 1.5. Since we require version 1.5, some code in res_resolver_unbound was also simplified. ASTERISK-28045 Reported by: Samuel Galarneau Change-Id: Iee94ad543cd6f8b118df8c4c7afd9c4e2ca1fa72	8 years ago
Corey Farrell	86ae100016	jansson: Backport fixes to bundled, use json_vsprintf if available. Use json_vsprintf from versions which contain fix for va_copy leak. Apply fixes from jansson master: * va_copy leak fix. * Avoid potential invalid memory read in json_pack. * Rename variable that shadowed another. Change-Id: I7522e462d2a52f53010ffa1e7d705c666ec35539	8 years ago
Kevin Harwell	720f76def0	rtp_engine: rtcp_report_to_json can overflow the ssrc integer value When writing an RTCP report to json the code attempts to pack the "ssrc" and "source_ssrc" unsigned integer values as a signed int value type. This of course means if the ssrc's unsigned value is greater than that which can fit into a signed integer value it gets converted to a negative number. Subsequently, the negative value goes out in the json report. This patch now packs the value as a json_int_t, which is the widest integer type available on a given system. This should make it so the value no longer overflows. Note, this was caught by two failing tests hep/rtcp-receiver/ and hep/rtcp-sender. Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0	8 years ago
George Joseph	80cc691252	stasis: Add function to delete topic from pool There's been a long standing leak when using topic pools. The topics in the pool get cleaned up when the last pool reference is released but you can't remove a topic specifically. If you reloaded app_voicemail for instance, and mailboxes went away, their topics were left in the pool. * Added stasis_topic_pool_delete_topic() so modules can clean up topics from pools. * Registered the topic pool containers so it can be examined from the CLI when AO2_DEBUG is enabled. They'll be named "<topic_pool_name>-pool". Change-Id: Ib7957951ee5c9b9b4482af7b9b4349112d62bc25	8 years ago
Joshua Colp	b5959113f6	pjproject: Upgrade to 2.8. This change brings in PJSIP 2.8, removes all the patches that were merged upstream, and makes a minor change to support a breaking change that was done. ASTERISK-28059 Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189	8 years ago
Sean Bright	5713fc0b49	autoconf: Check for srtp_get_version_string() before using it Change-Id: Id2a916ff9448706090e72ff2c7fb3f5ba24a05df	8 years ago
Sean Bright	5f6bef4e5b	res_pjsip: Log IPv6 addresses correctly Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name store IPv6 addresses without enclosing brackets. This causes some log output to be confusing because it is difficult to separate the IPv6 address from a port specification. * Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6 output. * When a pj_sockaddr is not available, explicitly wrap IPv6 addresses in brackets. * When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure to also set pjsip_rx_data.pkt_info.src_addr. Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8	8 years ago
Walter Doekes	4616c6508a	optional_api: Remove unused nonoptreq fields As they're not actively used, they only grow stale. The moduleinfo field itself is kept in Asterisk 13/15 for ABI compatibility. ASTERISK-28046 #close Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc	8 years ago
Sean Bright	9624cbff4f	res_pjproject: Add utility functions to convert between socket structures Currently, to convert from a pj_sockaddr to an ast_sockaddr, the address needs to be rendered to a string and then parsed into the correct structure. This also involves a call to getaddrinfo(3). The same is true for the inverse operation. Instead, because we know the internal structure of both ast_sockaddr and pj_sockaddr, we can translate directly between the two without the need for an intermediate string. Change-Id: If0fc4bba9643f755604c6ffbb0d7cc46020bc761	8 years ago
Joshua Colp	f233bba8e7	Merge "stasis: Reduce calculation of stasis message type hash." into 15	8 years ago
Joshua Colp	cba61db7a8	Merge "res_pjsip: Make pjlib.h consistently included." into 15	8 years ago
Richard Mudgett	45aeb08ada	res_pjsip.h: Fix doxygen comments. Change-Id: I9cf97bdc756012d1f552ab007f4aa85e0ddb4e62	8 years ago
Joshua Colp	78e6192262	stasis: Reduce calculation of stasis message type hash. When the stasis cache is used a hash is calculated for retrieving or inserting messages. This change calculates a hash when the message type is initialized that is then used each time needed. This ensures that the hash is calculated only once for the message type. Change-Id: I4fe6bfdafb55bf5c322dd313fbd8c32cce73ef37	8 years ago
Richard Mudgett	7e78539136	res_pjsip: Make pjlib.h consistently included. * Don't include pjlib.h twice in res_pjsip.h * Consistently use #include <> form for pjproject includes. (pjsip.h and pjlib.h) Change-Id: I3f7b42044840de64edf7e9d7695cb60c45990dc7	8 years ago
Alexander Traud	7fa8708460	BuildSystem: Enable Jansson in Solaris 11. In Solaris, the header <jansson.h> is in /usr/include/jansson. To find Jansson even in such a subdirectory, the tool pkg-config is queried via AST_PKG_CONFIG_CHECK. For those platforms, which do not list Jansson via pkg-config, the previous check remains and is executed thereafter. Because the check for the NetBSD Editline library uses the tool pkg-config conditionally PKG_PROG_PKG_CONFIG must be used. Because that check happens earlier than Jansson, it must be placed in front of that. The script configure does some pre-checks for the script configure of the Asterisk internal NetBSD Editline library. The check for the library ncurses should use not use the header <curses.h> but <ncurses.h>, because on some platforms <curses.h> is not a drop-in replacement for <ncurses.h>: For example in Solaris, the symbol initscr is a typedef in <curses.h> to a symbol which does not exist in the library ncurses (initscr32). Simply use <ncurses.h> when you link to ncurses. ASTERISK-27991 Change-Id: I69ea0f379f87a50049654b2487c76ee1c04fa53a	8 years ago
Joshua Colp	c52c915d7f	devicestate: Don't create topic when change isn't cached. When publishing a device state the change can be marked as being cachable or not. If it is not cached the change is just published to all interested and not stored away for later query. This was not fully taken into account when publishing in stasis. The act of publishing would create a topic for the device even if it may be ephemeral. This change makes it so messages which are not cached won't create a topic for the device. If a topic does already exist it will be published to but otherwise the change will only be published to the device state all topic. ASTERISK-27591 Change-Id: I18da0e8cbb18e79602e731020c46ba4101e59f0a	8 years ago
George Joseph	90f382b044	Merge "sched: Make ABI compatible between dev mode and non-dev mode." into 15	8 years ago
Joshua Colp	4fb3b1b090	sched: Make ABI compatible between dev mode and non-dev mode. In the past there was an assertion in the ast_sched_del function and in order to ensure it was useful the calling function name, line number, and filename had to be passed in. This cause the ABI to be different between dev mode and non-dev mode. This assertion is no longer present so the special logic can be removed to make it the same between them both. Change-Id: Icbc69c801e357d7004efc5cf2ab936d9b83b6ab8	8 years ago
George Joseph	9283685939	xmldoc.c: Fix dump of xml document The "xmldoc dump" cli command was simply concatenating xml documents into the output file. The resulting file had multiple "xml" processing instructions and multiple root elements which is illegal. Normally this isn't an issue because Asterisk has only 1 main xml documentation file but codec_opus has its own file so if it's downloaded and you do "xmldoc dump", the result is invalid. * Added 2 new functions to xml.c: ast_xml_copy_node_list creates a copy of a list of children. ast_xml_add_child_list adds a list to an existing list. * Modified handle_dump_docs to create a new output document and add to it the children from each input file. It then dumps the new document to the output file. Change-Id: I3f182d38c75776aee76413dadd2d489d54a85c07	8 years ago
George Joseph	104464b671	Merge "res_pjsip_sdp_rtp: include ice in ANSWER only if offered" into 15	8 years ago
Ben Ford	4048e25194	res_rtp_asterisk: Add support for sending NACK requests. Support has been added for receiving a NACK request and handling it. Now, Asterisk can detect when a NACK request should be sent and knows how to construct one based on the packets we've received from the remote end. A buffer has been added that will store out of order packets until we receive the packet we are expecting. Then, these packets are handled like normal and frames are queued to the core like normal. Asterisk knows which packets to request in the NACK request using a vector which stores the sequence numbers of the packets we are currently missing. If a missing packet is received, cycle through the buffer until we reach another packet we have not received yet. If the buffer reaches a certain size, send a NACK request. If the buffer reaches its max size, queue all frames to the core and wipe the buffer and vector. According to RFC3711, the NACK request must be sent out in a compound packet. All compound packets must start with a sender or receiver report, so some work was done to refactor the current sender / receiver code to allow it to be used without having to also include sdes information and automatically send the report. Also added additional functionality to ast_data_buffer, along with some testing. For more information, refer to the wiki page: https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements ASTERISK-27810 #close Change-Id: Idab644b08a1593659c92cda64132ccc203fe991d	8 years ago
Torrey Searle	7aef73bbc8	res_pjsip_sdp_rtp: include ice in ANSWER only if offered Keep track if ICE candidates were in the SDP offer & only put them in the corresponding SDP answer if the offer condaind ICE candidates ASTERISK-27957 #close Change-Id: Idf2597ee48e9a287e07aa4030bfa705430a13a92	8 years ago
George Joseph	4a23f39e3a	res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint A new option 'suppress_q850_reason_headers' has been added to the endpoint object. Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This option allows the 'Q.850' Reason header to be suppressed. The default value is 'no'. ASTERISK-27949 Reported-by: Ross Beer Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1	8 years ago
Richard Mudgett	7b63f32649	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
Jenkins2	ce6a661386	Merge "res_pjsip_session: Add ability to accept multiple sdp answers" into 15	8 years ago
George Joseph	541d343c20	res_pjsip_session: Add ability to accept multiple sdp answers pjproject by default currently will follow media forked during an INVITE on outbound calls if the To tag is different on a subsequent response as that on an earlier response. We handle this correctly. There have been reported cases where the To tag is the same but we still need to follow the media. The pjproject patch in this commit adds the capability to sip_inv and also adds the capability to control it at runtime. The original "different tag" behavior was always controllable at runtime but we never did anything with it and left it to default to TRUE. So, along with the pjproject patch, this commit adds options to both the system and endpoint objects to control the two behaviors, and a small logic change to session_inv_on_media_update in res_pjsip_session to control the behavior at the endpoint level. The default behavior for "different tags" remains the same at TRUE and the default for "same tag" is FALSE. Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6 ASTERISK-27936 Reported-by: Ross Beer	8 years ago
Alexander Traud	54a3340e3a	uuid: Enable UUID in Solaris 11. ASTERISK-27933 Reported by: bautsche Change-Id: I9b8362824efbfb2a16981e46e85f7c8322908c49	8 years ago
George Joseph	08b2d32e65	Merge "channel.c: Make CHECK_BLOCKING() save thread LWP id for messages." into 15	8 years ago
George Joseph	108f20a88a	Merge changes from topic 'ASTERISK-27625' into 15 * changes: channel.c: Fix usage of CHECK_BLOCKING() autoservice: Don't start channel autoservice if the thread is a user interface.	8 years ago
Jenkins2	31263f0ff6	Merge "ARI POST DTMF: Make not compete with channel's media thread." into 15	8 years ago
Jenkins2	d41b2a23e1	Merge "AMI PlayDTMF Action: Make not compete with channel's media thread." into 15	8 years ago
Kevin Harwell	6c90e0fb69	Merge "Fix some doxygen and curly placement." into 15	8 years ago
Richard Mudgett	4d598931ba	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	f82d0a9380	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	da2f32bd85	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	437fe5ac76	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	cf56ec739d	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	ce2aad08ee	Fix some doxygen and curly placement. Change-Id: I9a784a7c804120a8fa826c2a4cb9957e4b0b2fc8	8 years ago
Richard Mudgett	f0ebfa45b5	tcptls.h: Remove redundant SSL_CTX typedef. It is invalid to typedef something more than once. Though not all gcc compilers on different OS's complain about it. Change-Id: I5a7d4565990c985822d61ce75bde0b45f9870540	8 years ago
Alexander Traud	4b25cde9b3	res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. Furthermore, 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-27910 Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537	8 years ago
Alexander Traud	60242e1418	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
Joshua Colp	eb52c70f5b	Merge "rtp: Add support for RTP extension negotiation and abs-send-time." into 15	8 years ago
Joshua Colp	d94dd27667	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
Alexander Traud	ae2afa73a2	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
Jenkins2	2daabdeb03	Merge "rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code." into 15	8 years ago

1 2 3 4 5 ...

3935 Commits (15)