28473 Commits (7f8d570c8d03075c965b8154fdc24557f4492c47)

 

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Author	SHA1	Message	Date
zuul	d254f31426	Merge "chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP." into 14	9 years ago
zuul	f7b94dbf17	Merge "res_pjsip_registrar.c: Reduce stack usage in find_aor_name()." into 14	9 years ago
zuul	b378cc6f43	Merge "pjsip_configuration.c: Ignore repeated identify by methods." into 14	9 years ago
Joshua Colp	ed1106c891	Merge "config_global.c: Comments and a default expression adjustment." into 14	9 years ago
zuul	1c90758904	Merge "sip_to_pjsip.py: Map canreinvite as directmedia alias." into 14	9 years ago
Matt Jordan	827b235357	res/res_stasis_playback: Cancel the entire playlist when a stop occurs ... Prior to this patch, a stop issued by a delete of a Playback resource (indicated by the control frame AST_CONTROL_STREAM_STOP) would only stop the current media URI playing. Subsequent URIs specified by a playback operation would then proceed on, even though we had just indicated to the User that the Playback was finished *and* after they had just 'deleted' the resource. Whoops. This patch corrects it by bailing out of the sequence of URIs to play if one of them is terminated with an AST_CONTROL_STREAM_STOP indication. ASTERISK-26341 #close Change-Id: I2da9ec43545ba46cdfffe287c7e4907eae7fca42	9 years ago
zuul	045d29b2c0	Merge "sip_to_pjsip.py: Fix typo converting outboundproxy registration." into 14	9 years ago
zuul	4175b5081d	Merge "sip_to_pjsip.py: Fix comment typo and tabs." into 14	9 years ago
zuul	3f32149173	Merge "Sample configs: Eliminate false multiline comment block starts." into 14	9 years ago
Joshua Colp	e455036e9e	Merge "sorcery: Create function ast_sorcery_lockable_alloc." into 14	9 years ago
Joshua Colp	dec772d46c	Merge "named_locks: Use ao2_weakproxy to deal with cleanup from container." into 14	9 years ago
Joshua Colp	ddd6c810a9	Merge "astobj2: Support using a separate object for locking." into 14	9 years ago
George Joseph	8b6f9dde14	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
zuul	74a5bbfc1b	Merge "format_cap.c: Fix CLI "core show channeltype Surrogate" crash." into 14	9 years ago
Alexei Gradinari	09e9dca0c5	res_pjsip_session: 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 which could use the session's memory pools. If the session in the disconnected state then the session memory pools were already freed, so we get segfault. This patch adds a lifetime control on an INVITE session to pjproject. The lifetime of the session is manipulated by calling pjsip_inv_add_ref/pjsip_inv_dec_ref. This patch uses these functions to inform pjproject that the session is in use. This patch adds check if the session state is not disconnected and also checks if the memory pool is not NULL. This patch also places tasks 'session_end' and 'session_end_completion' into session's serializer to avoid race condition. ASTERISK-26291 #close Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7	9 years ago
Walter Doekes	9b59ebee2f	chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP. ... Certain SNOM phones send so-called "optional crypto" in their SDP body. Regular SRTP setup looks like this: m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... SNOM-style "optional crypto" looks like this: m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... A crypto line is supplied, but the m-line does not have SAVP. When res_srtp.so is *not* loaded, then chan_sip.so treats the optional crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the incoming call with the following message: WARNING: process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio For platforms that want to start providing SRTP this presents a compatibility problem. This changeset lets chan_sip handle the SDP as if no crypto-line was supplied: i.e. accept the call as regular RTP, just like it did before res_srtp was loaded. Now you'll get this informative warning instead: WARNING: Ignoring crypto attribute in SDP because RTP transport is insecure ASTERISK-23989 #close Reported by: Olle Johansson Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2	9 years ago
zuul	f33ccbdf86	Merge "app_mp3: Use correct buffer size and the same sample rate as the channel" into 14	9 years ago
Matt Jordan	3e054a3826	apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option ... In any scenario in which the callee is not connected to the caller, the current code in app_dial will crash due to raising a Dial End Stasis Message after the callee channel has been hung up. This patch corrects the error by simply moving the explicit hangup of the callee (peer) channel until after the dial end message. ASTERISK-25691 #close Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d	9 years ago
Matt Jordan	15105c7494	apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5 ... If the callee selects option '5' using the Dial application's privacy (P) option, the DIALSTATUS is erroneously set to ANSWER. This option reflects the callee sending the caller to VoiceMail one time; the call is definitely *not* ANSWERed in such a scenario. With this patch, the DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that is set when the 'send to VoiceMail every time' option is set. ASTERISK-25691 Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358	9 years ago
Richard Mudgett	323b74c6ca	res_pjsip_registrar.c: Reduce stack usage in find_aor_name(). ... Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09	9 years ago
Richard Mudgett	5bc8c91863	pjsip_configuration.c: Ignore repeated identify by methods. ... Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838	9 years ago
Richard Mudgett	bdfa4f8a0a	config_global.c: Comments and a default expression adjustment. ... Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3	9 years ago
Richard Mudgett	2c5bead338	sip_to_pjsip.py: Map canreinvite as directmedia alias. ... Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2	9 years ago
Richard Mudgett	d4f01fb7b6	sip_to_pjsip.py: Fix typo converting outboundproxy registration. ... Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15	9 years ago
Richard Mudgett	4ffc59b4d5	sip_to_pjsip.py: Fix comment typo and tabs. ... Change-Id: If35174614545727817d329c60ba4456c028941b5	9 years ago
Richard Mudgett	bc5b3a7217	Sample configs: Eliminate false multiline comment block starts. ... Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6	9 years ago
Richard Mudgett	2a8d27e020	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	e65f170d9c	sorcery: Create function ast_sorcery_lockable_alloc. ... Create an alternative to ast_sorcery_generic_alloc which uses astobj2 shared locking. Use this new method for the 'struct ast_sip_aor' allocator. Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f	9 years ago
Corey Farrell	3af3860151	named_locks: Use ao2_weakproxy to deal with cleanup from container. ... This allows standard ao2 functions to be used to release references to an ast_named_lock. This change can cause less frequent locking of the global named_locks container. The container is no longer locked when a named_lock reference is being release except when this causes the named_lock to be destroyed. Change-Id: I644e39c6d83a153d71b3fae77ec05599d725e7e6	9 years ago
Corey Farrell	e260e507f2	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	971c96e3df	Merge "res_pjsip: qualify/unqualify added/deleted realtime endpoints" into 14	9 years ago
zuul	05a67b0b3f	Merge "sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations." into 14	9 years ago
Michael Kuron	dd408708c9	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
zuul	7402b7ceb4	Merge "followme: initialize all config items on reload" into 14	9 years ago
Alexei Gradinari	411efb9754	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	a220f40cfe	Merge "res_pjsip: Default endpoints to the "offline" status." into 14	9 years ago
Joshua Colp	a804abbe99	Merge "pjproject_bundled: Disable srtp use by pjmedia" into 14	9 years ago
zuul	2fc674129e	Merge "pbx.c: Prevent infinite recursion in manager_show_dialplan_helper." into 14	9 years ago
zuul	cb176619fe	Merge "app_queue: Ensure member is removed from pending when hanging up." into 14	9 years ago
Mark Michelson	4c2fed8814	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
chrisderock	a637e2c301	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
Etienne Lessard	f9dc20b8cb	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	f7b3f628cd	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	9dfd4916a8	Merge "res_pjsip: Cache global config options." into 14	9 years ago
zuul	4a2371f1cd	Merge "channel: No hung-up on failing security requirements." into 14	9 years ago
George Joseph	cfe2675dd4	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
Alexander Traud	1d2d4e2ae9	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	c3ac1651dd	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	a8771d7a71	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	1a7d5989d6	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