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Author	SHA1	Message	Date
Joshua Colp	24d0907849	Merge "pjsip: Fix a few media bugs with reinvites and asymmetric payloads."	9 years ago
zuul	6f73da9c11	Merge "app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS."	9 years ago
Tzafrir Cohen	0646b48ece	chan_dahdi: remove by_name support ... Support for referring to DAHDI channels by logical names was added in (FIXME: when? Asterisk 11? 1.8?) and was intended to be part of support of refering to channels by name. While technically usable, it has never been properly supported in dahdi-tools, as using it would require many changes at the Asterisk level. Instead logical mapping was added at the kernel level. Thus it seems that refering to DAHDI channels by name is not really used by anyone, and therefore should probably be removed. Change-Id: I7d50bbfd9d957586f5cd06570244ef87bd54b485	9 years ago
zuul	90d1b3262b	Merge "res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls."	9 years ago
George Joseph	4f45d62653	pjproject_bundled: Remove usage of tar's --strip-components option ... Older versions of tar don't support the --strip-components option so instead of doing 'tar --strip-components=1 -C source', we now just untar to the tarball's root directory (pjproject-<version>) and rename that directory to 'source'. Also fixed an issue where the pjproject source directory is a hard coded absolute pathname. ASTERISK-26510 #close ASTERISK-22480 #close Change-Id: I9ec92952507a91ff4e4d01e0149e09fd8e8f32b0	9 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
Joshua Colp	6993f3c9c3	res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls. ... The res_pjsip_caller_id module wrongly assumed that a saved From header would always exist on sessions. This is true until an inbound call is received and a session timer causes an UPDATE to be sent. In this case there will be no saved From header and a crash will occur. This change makes it fall back to the From header of the outgoing request if no saved From header is present. ASTERISK-26307 #close Change-Id: Iccc3bc8d243b5ede9b81abf960292930c908d4fa	9 years ago
zuul	211a1003e4	Merge "test_astobj2_thrash: Fix multithreaded issues"	9 years ago
zuul	761b32633c	Merge "cdr_radius,cel_radius: Fix old memleak in unload"	9 years ago
Joshua Colp	95062fe220	app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS. ... When executing the MailboxExists dialplan application and MAILBOX_EXISTS dialplan function the passed in temporary voice mailbox was not cleared, causing it to try to free garbage. ASTERISK-26503 #close Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3	9 years ago
Joshua Colp	aed6c219a3	pjsip: Fix a few media bugs with reinvites and asymmetric payloads. ... When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc	9 years ago
Joshua Colp	7925f60cd9	res_pjsip_sdp_rtp: Fix address family of explicit media_address. ... When an explicit media_address is provided the address family in the SDP needs to be set to reflect it. ASTERISK-26309 Change-Id: Ib9350cc91c120eb2f96f0623d3907d12af67eb79	9 years ago
George Joseph	802bbf8752	test_astobj2_thrash: Fix multithreaded issues ... The test uses 4 threads to grow, count, lookup and shrink 15K objects in a container. If there's only 1 execution engine available, the test will complete in <50ms. If each threads gets its own execution engine, the test may timeout after 60 seconds because the count thread does a locked ao2_callback on the whole container in a tight loop with only a sched_yield to give up time. The lock contention makes the test execution times wildly variable and mostly timeout. 2 execution engines are OK, 3 results in about 33% failure rate and >=4 causes a 80% failure rate. To fix, the sched_yield was changed to a usleep(500). Also, the number of buckets specified for the container was an even number so that was changed to the next prime number greater than (MAX_HASH_ENTRIES / 100). That's 151 currently. Change-Id: I50cd2344161ea61bfe4b96d2a29a6ccf88385c77	9 years ago
Alexei Gradinari	2b9ad3a5f7	chan_pjsip: segfault on already disconnected session ... On heavy loaded system the TCP/TLS incoming calls could be disconnected by pjproject while these calls are being processed by asterisk. This patch uses functions pjsip_inv_add_ref/pjsip_inv_dec_ref to inform pjproject that an INVITE session is in use. ASTERISK-26482 #close Change-Id: Ia2e3e2f75358cdb530252a9ce158af3d5d9fdf33	9 years ago
Badalyan Vyacheslav	01d1d3763f	cdr_radius,cel_radius: Fix old memleak in unload ... - Call "rc_openlog" optional. If you do not call, you will simply NULL instead of a name. - On the one PID can be only one syslog channel. And it can already be run in logger.c - Calling rc_openlog we assigns a new name for the channel syslog. This unexpected behavior for logger.c. Most lesser evil, is to agree on a NULL name syslog if the channel was not launched in logger.c. It also solves the problem of memory leaks. ASTERISK-26455 #close Change-Id: Ic17c38de67583e971d78fe18807d1a9faf8f0afd	9 years ago
Joshua Colp	7c79b057dd	Merge "pjsip: Support dual stack automatically."	9 years ago
Joshua Colp	3972ab2e19	Merge "pjproject_bundled: Fixed various build issues"	9 years ago
Joshua Colp	0af69e4454	Merge "ARI: Add duplicate channel ID checking for channel creation."	9 years ago
Joshua Colp	51ae1046d0	Merge "ARI: Detect duplicate channel IDs"	9 years ago
George Joseph	16c23b57c7	pjproject_bundled: Fixed various build issues ... * CFLAGS is now properly set when using older gcc. * All third-party pjproject targets have been removed. This fixes an issue with older libsrtp in some distros. * Manually removing the source directory now causes a rebuild. * EXTERNALS_CACHE_DIR is now properly checked. * Whitespace fixes. Change-Id: I98fec6847efc5602a9f41cb95096fd660a49fa60	9 years ago
Pascal Cadotte Michaud	1d277e7cb6	typo: s/paranthesis/parenthesis/ in a comment ... Change-Id: I7c1f4eb051177ee22cbe97e063d4a3effe29be30	9 years ago
Joshua Colp	403c4f5833	pjsip: Support dual stack automatically. ... This change adds support for dual stack automatically. No configuration is required and the IP address and version in the SIP messages and SDP will be automatically changed based on the transport over which the message is being sent. RTP usage has also been changed to listen on both IPv4 and IPv6 simultaneously to allow media to flow, and to allow ICE support on both simultaneously. This also allows failover between IPv6 and IPv4 to work as expected. ASTERISK-26309 #close Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d	9 years ago
Mark Michelson	3bd76dd679	ARI: Add duplicate channel ID checking for channel creation. ... This is similar to what is done for origination, but for the 14 and up channel creation method. When attempting to create a channel, if a channel ID is specified and a channel already exists with that ID, then a 409 is returned. Change-Id: I77f9253278c6947939c418073b6b31065489187c	9 years ago
Mark Michelson	e459b8dadf	ARI: Detect duplicate channel IDs ... ARI and AMI allow for an explicit channel ID to be specified when originating channels. Unfortunately, there is nothing in place to prevent someone from using the same ID for multiple channels. Further complicating things, adding ID validation to channel allocation makes it impossible for ARI to discern why channel allocation failed, resulting in a vague error code being returned. The fix for this is to institute a new method for channel errors to be discerned. The method mirrors errno, in that when an error occurs, the caller can consult the channel errno value to determine what the error was. This initial iteration of the feature only introduces "unknown" and "channel ID exists" errors. However, it's possible to add more errors as needed. ARI uses this feature to determine why channel allocation failed and can return a 409 error during origination to show that a channel with the given ID already exists. ASTERISK-26421 Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06	9 years ago
snuffy	e03364c40a	Fix issue with CLI not returning to prompt after running "features show" ... ASTERISK-26444 #close Change-Id: I91d645b7e6e5dba35f8c410df2be77a8c0e3acb8	9 years ago
zuul	7c00b34372	Merge "utils.c: Fix ast_set_default_eid for multiple platforms"	9 years ago
zuul	bc5fdbd10a	Merge "res_rtp_asterisk: Add ice_blacklist option"	9 years ago
Joshua Colp	1843b7fa0c	Merge "chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia."	9 years ago
Michael Walton	3e96d491d0	res_rtp_asterisk: Add ice_blacklist option ... Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the form ice_blacklist = <subnet spec>, e.g. ice_blacklist = 192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay discovery. This is useful for optimizing the ICE process where a system has multiple host address ranges and/or physical interfaces and certain of them are not expected to be used for RTP. Multiple ice_blacklist configuration lines may be used. If left unconfigured, all discovered host addresses are used, as per previous behavior. Documention in rtp.conf.sample. ASTERISK-26418 #close Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9	9 years ago
Mark Michelson	f14ef51ead	CDR: Alter destruction pattern for CDR chains. ... CDRs form chains. When the root of the chain is destroyed, it then unreferences the next CDR in the chain. That CDR is destroyed, and it then unreferences the next CDR in the chain. This repeats until the end of the chain is reached. While this typically does not cause any sort of problems, it is possible in strange scenarios for the CDR chain to grow way longer than expected. In such a scenario, the destruction pattern can result in a stack overflow. This patch fixes the problem by switching from a recursive pattern to an iterative pattern for destruction. When the root CDR is destroyed, it is responsible for iterating over the rest of the CDRs and unreferencing each one. Other CDRs in the chain, since they are not the root, will simply destroy themselves and be done. This causes the stack depth not to increase. ASTERISK-26421 #close Reported by Andrew Nagy Change-Id: I3ca90c2b8051f3b7ead2e0e43f60d2c18fb204b8	9 years ago
zuul	193a94a60f	Merge "menuselect: invalid test for GTK2"	9 years ago
Joshua Colp	d1bbef094c	Merge "cli: Auto-complete File not Module for core set debug."	9 years ago
Joshua Colp	f31772ec20	ari: Update model validator based on addition of asterisk_id. ... ASTERISK-26470 Change-Id: I9c386f7a1c7d969161b28f189eb6298bbc5b7541	9 years ago
Joshua Colp	e29e3a4f11	Merge "Binaural synthesis (confbridge): On/off setting for binaural synthesis."	9 years ago
Joshua Colp	8b2b8be4e3	Merge "chan_rtp: Set a sane default rtp engine for unicast."	9 years ago
Tzafrir Cohen	18a6f250e2	menuselect: invalid test for GTK2 ... configuire.ac was only checking for the existence of pkg-config and not the gtk2 package itself. Now it calls AST_PKG_CONFIG_CHECK for gtk+-2.0. ASTERISK-26356 #close Change-Id: I93e9d0166341f0e7f84b52955bb6f81da42f2ef6	9 years ago
Joshua Colp	9cd3254b98	Merge "res/ari: Add the Asterisk EID field to outgoing events"	9 years ago
Alexander Traud	a43ee21211	cli: Auto-complete File not Module for core set debug. ... Since Asterisk 1.8, the command "core set debug" on the command-line interface asks not for a file (.c) but a module name. This change shows modules (.so) on the auto-completion via a tabulator or the question mark. Now, when you partially type a module name, TAB or ?, you get the correct candidiates. ASTERISK-26480 Change-Id: I1213f1dd409bd4ff8de08ad80cb0c73cafb1bae0	9 years ago
zuul	a1c62c930a	Merge "app_queue: Added initialization for "context" parameter"	9 years ago
frahaase	dce31f90ba	Binaural synthesis (confbridge): On/off setting for binaural synthesis. ... Adds setting to confbridge.conf (binaural_active) that determines if binaural synthesis can be available in bridge_softmix. ASTERISK-26292 Change-Id: I59dfcb8e55fe1df4ef32045882fea5bb58fc71db	9 years ago
George Joseph	2a808b2fa6	pjproject_bundled: Add patch to address SSL crash ... Addresses crashes when an attempt is made to operate on an SSL socket after the socket has been closed. ASTERISK-26477 #close Change-Id: I421305b357558b4f9e690210dc0f4831ef4b3002	9 years ago
Leandro Dardini	973e57d5ce	app_queue: Added initialization for "context" parameter ... When using Asterisk Realtime Architecture, empty fields are skipped and the default values are used. If the "context" parameter in queue was set and then cleared from the database, the old value remains in memory and it continues to be used. This change initialize the "context" parameter with an empty value, allowing clearing the parameter. ASTERISK-26462 #close Change-Id: I64be73d5044ce38dd02408bd0e53de965ef65905	9 years ago
Matt Jordan	dd5129d84a	res/ari: Add the Asterisk EID field to outgoing events ... This patch adds the Asterisk EID field to all outgoing ARI events. Because this field should be added to all events as they are transmitted, it is appended to the JSON message just prior to it being handed off to the application message handler. This makes it somewhat resilient to both new events being added to ARI, as well as other potential event transport mechanisms. ASTERISK-26470 #close Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d	9 years ago
Moises Silva	2b03017022	chan_rtp: Set a sane default rtp engine for unicast. ... ASTERISK-26439 Change-Id: I7f5ee2eeba8906e9ecb3293dbe3a747770bb5011	9 years ago
George Joseph	6651c66e68	utils.c: Fix ast_set_default_eid for multiple platforms ... ast_set_default_eid was searching for ethX, emX, enoX, ensX and even pciD#U interface names. While this was a good attempt, it wasn't inclusive enough to capture interfaces like enp6s0 or ens6d1, etc. Rather than relying on interface names, we now simply find the first interface returned by the OS that has a hardware address and that address isn't all 0x00 or all 0xff. The code IS different for BSD, Solaris and Linux based on what method is available for enumerating interfaces. Tested on: FreeBSD9 CentOS6 Ubuntu14 Fedora24 I was unable to test on Solaris at this time but the code for Solaris is used elsewhere at Digium. Change-Id: Iaa6db87ca78a9a375e47d70e043ae08c1448cb72	9 years ago
Michael Kuron	e9315791b3	chan_sip: Only send video on outgoing channel if incoming channel supports it ... Previously, the settings videosupport=always and videosupport=yes behaved identically and unconditionally caused a video offer to be sent in the SDP on an outgoing call. This was a regression introduced with commit 5a1d90e1fb in Asterisk 1.6.1. This commit restores correct behavior: videosupport=always causes a video offer to be sent unconditionally, while videosupport=yes will only offer video on an outbound channel if the incoming channel it is bridged to also supports video. That way, the device receiving the outgoing call can display the correct user interface elements for audio or video and will not unnecessarily show a blank video window on an audio-only call. ASTERISK-17470 #close Change-Id: I782f4409d436114dbc97061c3570c0cd24f7c3ae	9 years ago
zuul	5df575e16a	Merge "Fix issues with bundled pjproject cached download."	9 years ago
zuul	6f49ca5927	Merge "Audit ast_json_pack() calls for needed UTF-8 checks."	9 years ago
zuul	9581407fd2	Merge "json: Check party id name, number, subaddresses for UTF-8."	9 years ago
zuul	5b70ae04d4	Merge "json: Add UTF-8 check call."	9 years ago