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Author	SHA1	Message	Date
George Joseph	6caf6bcdad	build: Add download capability for external packages ... The DPMA and g729a, silk, siren7 and siren14 codecs hosted at http://downloads.digium.com/pub/telephony/ are now listed in the "External" sections of the "Resource Modules" and "Codec Translators" pages in menuselect. Any that are selected will automatically be downloaded and installed when "make install" is run. Their LICENSE and README (if avaialble) files will be installed to ASTVARLIBDIR/documentation/thirdparty/<product_name>. Example use with codecs: The codecs/codecs.xml file is a menuselect style xml file that lists the codecs to be included. Their support levels are 'external', which triggers the download and install, and defaultenabled is no. Also because codec_g729a is actually in a directory named codec_g729 on the download server, the newly added 'member_data' element is used to override the default of the directory name being the package name. You can use the 'directory_name' attribute to keep default base URL (http://downloads.digium.com/pub/telephony/) but use the new directory, or you use the 'remote_url' attribute to specify a full URL to the download directory. In this case, you must still follow the same subdirectory naming conventions as that used for the packages located at 'http://downloads.digium.com/pub/telephony'. A new configure option '--with-externals-cache' was added and like '--with-sounds-cache' it allows the installer to cache tarballs so they're not downloaded every time. To assist with the download and install process, each external package now has a manifest.xml file that, among other things, contains a package version and checksums for each file in the tarball. The manifest is saved to both the cache directory and ASTMODDIR and together with the manifest.xml on the downloads site, tells the install scripts whether a download and/or update is needed. bash and xmlstarlet are required for downloader operation. If they're not installed, the external items in menuselect will be unavailable. Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a	9 years ago
Joshua Colp	fe806ba08b	Merge "format_cap.c: Fix CLI "core show channeltype Surrogate" crash."	9 years ago
zuul	d57242a16b	Merge "astobj2: Support using a separate object for locking."	9 years ago
Joshua Colp	e34f299a96	Merge "codecs: Add Codec 2 mode 2400."	9 years ago
zuul	f87008f11a	Merge "app_mp3: Use correct buffer size and the same sample rate as the channel"	9 years ago
Richard Mudgett	c3b965a2c0	format_cap.c: Fix CLI "core show channeltype Surrogate" crash. ... * Make ast_format_cap_get_names() NULL tolerant. ASTERISK-26331 #close Reported by: CGI.NET Change-Id: Id67e93936dc8ec2a33a9d33655843d43b59285a3	9 years ago
Corey Farrell	0c5b6e9ff5	astobj2: Support using a separate object for locking. ... Create ao2_alloc_with_lockobj function to support shared locking. Change-Id: Iba687eb9843922be7e481e23a32c0700ecf88a80	9 years ago
zuul	d3c4b901d4	Merge "res_pjsip: qualify/unqualify added/deleted realtime endpoints"	9 years ago
Joshua Colp	cc26efece3	Merge "sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations."	9 years ago
Michael Kuron	48fd4c815c	app_mp3: Use correct buffer size and the same sample rate as the channel ... Previously, the buffer used for MP3 streamed from HTTP servers had a size of 1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1 minute. Only when the buffer is full does audio start to play. For MP3 files streamed from a server, that is usually not a big deal as long as the connection to the server is fast enough to supply that much data within a second or two. For MP3 live streams however, it takes 1 minute to download 1 minute of audio, so without this change, app_mp3 wasn't really usable for MP3 live streams. This commit changes the buffer size so that it covers 6 seconds of an MP3 file streamed from a server and 0.5 seconds of an MP3 live stream. The latter is identified by the use of a .m3u file extension. app_mp3 so far only supported 8 kHz audio. Now it always runs at the sample rate of the channel. ASTERISK-26085 #close Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0	9 years ago
Alexei Gradinari	faf9bdebb7	res_pjsip: qualify/unqualify added/deleted realtime endpoints ... If the PJSIP endpoint's AOR with the permanent contact was deleted from the realtime storage the res_pjsip module continues trying to qualify this contact. The error 'Unable to find an endpoint to qualify contact' appeares every 'qualify_frequency' seconds. This patch deletes this contact in this case. The PJSIP endpoint's AOR with the permanent contact is never qualified if it is added to realtime storage after asterisk started. This patch adds qualifying for the AOR's permanent contacts on the first handling of this AOR. ASTERISK-26319 #close Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe	9 years ago
zuul	e7d06a8097	Merge "res_pjsip: Default endpoints to the "offline" status."	9 years ago
zuul	e91fc62f80	Merge "pjproject_bundled: Disable srtp use by pjmedia"	9 years ago
zuul	b869bf0f38	Merge "pbx.c: Prevent infinite recursion in manager_show_dialplan_helper."	9 years ago
zuul	8bdd5b63df	Merge "app_queue: Ensure member is removed from pending when hanging up."	9 years ago
zuul	b0b480592a	Merge "app_macro: Consider '~~s~~' as a macro start extension."	9 years ago
Mark Michelson	c98a047ee6	res_pjsip: Default endpoints to the "offline" status. ... A recent change attempted to optimize startup by not updating contact status. Instead, code responsible for qualifying contacts updates the status as it becomes known. The code even accounts for contacts/AORs that are not set to be qualified. The problem, though, is when there are no contacts associated with an endpoint. A common case is when an endpoint is set to register its contacts but has not done so yet. In this case, prior to registration, the endpoint's device state will appear to be "not in use" and hints associated with that device will appear to be "idle". In actuality, the device state and hint should both appear as "unavailable". The reason for the failure is that the optimization change made all persistent endpoint states set to "unknown". The fix here is to change the hard-coded "unknown" to be "offline" instead. The default state will be offline until the qualifying code determines that the contact is actually online. This way, if there are no contacts at all, then the state stays as offline, and device state and hints appear correctly. ASTERISK-26269 #close Reported by nappsoft Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a	9 years ago
Etienne Lessard	5e0758575c	pbx.c: Prevent infinite recursion in manager_show_dialplan_helper. ... Previously, if context A was including context B and context B was including context A, i.e. if there was a circular dependency between contexts, then calling manager_show_dialplan_helper could lead to an infinite recursion, resulting in a crash. This commit applies the same solution as the one implemented in the show_dialplan_helper function. The manager_show_dialplan_helper and show_dialplan_helper functions contain lots of code in common, but the former was missing the "infinite recursion avoidance" code. ASTERISK-26226 #close Change-Id: I1aea85133c21787226f4f8442253a93000aa0897	9 years ago
Joshua Colp	c21e6764f1	app_queue: Ensure member is removed from pending when hanging up. ... When dialing channels it is possible that they may not ever leave the not in use state (Local channels in particular) by the time we cancel them. If this occurs but we know they were dialed we explicitly remove them from the pending members container so that subsequent call attempts occur. ASTERISK-26299 #close Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65	9 years ago
zuul	90b7f7fdb5	Merge "res_pjsip: Cache global config options."	9 years ago
zuul	4d06f4621a	Merge "channel: No hung-up on failing security requirements."	9 years ago
George Joseph	a7487e9261	pjproject_bundled: Disable srtp use by pjmedia ... The reason for the disable is that while Asterisk works fine with older libsrtp versions, newer versions of pjproject won't compile with them. Debian 6 for instance, has libsrtp 1.4.4 which is older than what pjproject is expecting. We don't use most of pjmedia but we DO use it for SDP negotiation. Luckily disabling srtp in pjmedia doesn't interfere with it's ability to negitiate a secure channel. The proper crypto attributes are negotiated in both directions. ASTERISK-26279 #close Change-Id: Id25a92cdf3df97a26c53cffae65b6b82de33c8e2	9 years ago
Joshua Colp	4a8bdfc49b	Merge "res_fax: Fix deadlock in ast_channel_get_t38_state()."	9 years ago
Joshua Colp	179e8c15c8	Merge "res_fax: Fix deadlock setting FAXMODE channel variable."	9 years ago
Joshua Colp	383b35fca7	Merge "res_fax.c: Fix deadlock in fax_gateway_indicate_t38()."	9 years ago
Joshua Colp	25e9356bb9	Merge "res_fax.c: Add chan locked precondition comments."	9 years ago
Joshua Colp	44b8cc8b48	Merge "ast_framehook_detach() must be called with the channel locked."	9 years ago
zuul	795532b2d5	Merge "ast_framehook_attach() must be called with the channel locked."	9 years ago
zuul	c82cef8441	Merge "Fix checks for allocation debugging."	9 years ago
zuul	e3e08e1131	Merge "Fix naming mismatch of allocator functions."	9 years ago
Alexander Traud	858fa5eb2c	channel: No hung-up on failing security requirements. ... In your Diaplan, if you specify same => n,Set(CHANNEL(secure_bridge_media)=1) same => n,Set(CHANNEL(secure_bridge_signaling)=1) only the SIP channel driver chan_sip supports this. All other channels drivers like res_pjsip fail. In case of failure, the original sRTP source code released the whole channel, even if not hung-up, yet. This change does not release the channel but instead hangs-up the channel. ASTERISK-26306 Change-Id: I0489f0cb660fab6673b0db8af027d116e70a66db	9 years ago
Alexander Traud	f35501b8c9	sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations. ... When using the migration script sip_to_pjsip.py, and your sip.conf is configured with bindaddr=::, two transports are written to pjsip.conf, one for 0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4 and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface like in chan_sip. Furthermore, the script internal functions "build_host" and "split_hostport" did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change makes sure, even such addresses are parsed correctly. ASTERISK-26309 Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48	9 years ago
Richard Mudgett	ea929d766d	res_pjsip: Cache global config options. ... We may check a global config option hundreds of times a second or more. Asking sorcery for the global configuration from the config files backend involves several allocations and container traversals. Using realtime without a memory cache is a lot worse because you have to lookup in the realtime database each time to reconstitute the sorcery object. With a memory cache for realtime, there is about the same amount of overhead as for config files. Either way, it is still fairly expensive to access the sorcery object that much. * Cache the global config options so we can access them faster. You must now always perform a res_pjsip reload to change the global options. Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7	9 years ago
Richard Mudgett	5eb6cb969f	res_fax: Fix deadlock in ast_channel_get_t38_state(). ... ast_channel_get_t38_state() calls ast_channel_queryoption() with AST_OPTION_T38_STATE. If the passed in channel is a local channel then a deadlock can happen if a channel lock is held when called. * Made ast_channel_get_t38_state() callers not hold a channel lock before calling. * Update ast_channel_get_t38_state() doxygen to note that no channel locks can be held when calling the function. ASTERISK-26203 #close Reported by: Etienne Lessard ASTERISK-24822 #close Reported by: David Brillert ASTERISK-22732 #close Reported by: Richard Mudgett Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214	9 years ago
Richard Mudgett	277a2d667a	res_fax: Fix deadlock setting FAXMODE channel variable. ... ASTERISK-25980 added the FAXMODE channel variable to res_fax.c. Unfortunately, it also introduced a deadlock potential because set_channel_variables() which sets FAXMODE can be called during a masquerade. The ast_channel_get_t38_state() which gets the value used to set FAXMODE cannot be called with the channel locked. As a result, local channels can deadlock because of how they must acquire the locks necessary to operate. The intent of FAXMODE is for dialplan to know how a fax was transferred after the fax completes. However, the previous patch sets FAXMODE to the channel's current T.38 state AFTER the fax has completed and where T.38 may have already disconnected. * Set FAXMODE based upon T.38 negotiations exchanged either with the fax applications or the fax framehooks. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1	9 years ago
Richard Mudgett	edca14c8a5	res_fax.c: Fix deadlock in fax_gateway_indicate_t38(). ... fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be called with any channel locks already held. A deadlock can happen if the function is operating on a local channel. * Made fax_gateway_indicate_t38() unlock the channel before calling ast_indicate_data() since fax_gateway_indicate_t38() is always called with the channel locked. * Made fax_gateway_indicate_t38() return void since nothing cared about its return value. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407	9 years ago
Richard Mudgett	141cd42880	res_fax.c: Add chan locked precondition comments. ... Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7	9 years ago
Richard Mudgett	b86771d1bf	ast_framehook_detach() must be called with the channel locked. ... The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584	9 years ago
Richard Mudgett	5744f434f0	ast_framehook_attach() must be called with the channel locked. ... The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438	9 years ago
chrisderock	93b7533d74	app_macro: Consider '~~s~~' as a macro start extension. ... As described in issue ASTERISK-26282 the AEL parser creates macros with extension '~~s~~'. app_macro searches only for extension 's' so the created extension cannot be found. with this patch app_macro searches for both extensions and performs the right extension. ASTERISK-26282 #close Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb	9 years ago
varnav	d2e03c252d	chan_iax2: Set plaintext auth to deprecated as per ASTERISK-22820 ... Starting from draft 2 of RFC 5456 (October 23, 2006) plaintext auth is not supported in IAX2 protocol. Please refer to section 8.6.13 of RFC 5456. But plaintext auth is still supported by Asterisk implementation of IAX2. This support should be dropped. Patch, based on asterisk-dev discussion, adds deprecation warning on startup if 'auth' is set to 'plaintext', changes default values of 'auth' from 'md5, plaintext' to 'md5'. Patch is safe in terms of backwards compatibility, will work even if remote peers have auth=plaintext and we have defaults. auth=plaintext setting will remain deprecated in Asterisk 14 and 15, and IAX2 plaintext support will be removed in Asterisk 16. ASTERISK-22820 #close Change-Id: I5d2f3830cb57645604818f87518916e8a5c317bf	9 years ago
George Joseph	e40aa40aca	res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_options ... ast_multicast_rtp_create_options now checks for NULL or empty options Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362	9 years ago
Alexander Traud	2e79f52d71	codecs: Add Codec 2 mode 2400. ... ASTERISK-26217 #close Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6	9 years ago
Mark Michelson	ded22c712a	ConfBridge: Rework announcer channel methodology ... NOTE: This patch was submitted earlier and reverted because of a failing test. The test has been patched so that it adjusts for the changes here, so this is being resubmitted for review. One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0	9 years ago
Joshua Colp	11ef7f34bf	Merge "Revert "ConfBridge: Rework announcer channel methodology""	9 years ago
Joshua Colp	065d810d3f	Revert "ConfBridge: Rework announcer channel methodology" ... This reverts commit 5aa8773052. Change-Id: I9ab45776e54a54ecf1bac9ae62d976dec30ef491	9 years ago
zuul	c9df806f24	Merge "ConfBridge: Rework announcer channel methodology"	9 years ago
zuul	4913fe3825	Merge "followme: initialize all config items on reload"	9 years ago
zuul	27813c7439	Merge "compilation failed with -Werror=maybe-uninitialized"	9 years ago
zuul	47c9acb5b2	Merge "res_odbc_transaction: add dep on generic_odbc"	9 years ago