asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	edaba05fea	Merge "build: Add download capability for external packages"	9 years ago
Alexander Traud	7a12355dbd	chan_sip: Allow Preferred sRTP. Following the Encrypt-all-the-things paradigm: The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile (sRTP is preferred aka optional; not mandatory). If the VoIP server does not support sRTP and TLS, the phone shows an open padlock icon. This paradigm is supported by several VoIP/SIP clients on default. Some implementations even cannot be changed to RTP/sAVP. Therefore here, this change allows Preferred sRTP for ingress. For egress, please, create a dial plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP. ASTERISK-20234 #close Reported by: tootai Tested by: tootai, Alexander Traud patches: srtp_patches.diff submitted by Matt Jordan Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd	9 years ago
Joshua Colp	e6cad17d6d	Merge "chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP."	9 years ago
Joshua Colp	2ff853279f	Merge "pjsip_configuration.c: Ignore repeated identify by methods."	9 years ago
zuul	43ef73ad45	Merge "resource_channels.c: add hangup reason "answered_elsewhere"."	9 years ago
zuul	7437467d94	Merge "res_pjsip_registrar.c: Reduce stack usage in find_aor_name()."	9 years ago
zuul	d0beb475b4	Merge "config_global.c: Comments and a default expression adjustment."	9 years ago
zuul	05240e2b57	Merge "sip_to_pjsip.py: Map canreinvite as directmedia alias."	9 years ago
zuul	eae37c3524	Merge "sip_to_pjsip.py: Fix typo converting outboundproxy registration."	9 years ago
zuul	eac6eef4ec	Merge "sip_to_pjsip.py: Fix comment typo and tabs."	9 years ago
zuul	5fb547a9ca	Merge "Sample configs: Eliminate false multiline comment block starts."	9 years ago
zuul	b5e4445b29	Merge "sorcery: Create function ast_sorcery_lockable_alloc."	9 years ago
zuul	825d6e036c	Merge "named_locks: Use ao2_weakproxy to deal with cleanup from container."	9 years ago
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
Walter Doekes	d80b28560c	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
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	68c7694abb	res_pjsip_registrar.c: Reduce stack usage in find_aor_name(). Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09	9 years ago
Richard Mudgett	35ce4d25c7	pjsip_configuration.c: Ignore repeated identify by methods. Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838	9 years ago
Richard Mudgett	c1e438fdf7	config_global.c: Comments and a default expression adjustment. Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3	9 years ago
Richard Mudgett	edcf09e47c	sip_to_pjsip.py: Map canreinvite as directmedia alias. Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2	9 years ago
Richard Mudgett	47336a0bdd	sip_to_pjsip.py: Fix typo converting outboundproxy registration. Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15	9 years ago
Richard Mudgett	dba02575fc	sip_to_pjsip.py: Fix comment typo and tabs. Change-Id: If35174614545727817d329c60ba4456c028941b5	9 years ago
Richard Mudgett	4aaa27e532	Sample configs: Eliminate false multiline comment block starts. Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6	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	e875e1c12a	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	131baf70d6	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	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
Jean Aunis	91993ebaa5	resource_channels.c: add hangup reason "answered_elsewhere". In ARI, the channels API allows to hangup a channel with a hangup reason. This commit adds a new reason "answered_elsewhere". When using a SIP channel, this will eventually allow Asterisk to add a proper "Reason" header to a CANCEL message. ASTERISK-26321 Change-Id: Ia97675bd4acd6a7f58eb467953dfb94559f6583d	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

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