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2016-09-09res_pjsip: Add ignore_uri_user_options option.Richard Mudgett
This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
2016-09-09Merge "res/res_pjsip: Add preferred_codec_only config to pjsip endpoint."zuul
2016-09-09res_pjsip: Only invoke unidentified endpoint logic when unidentified.Joshua Colp
The code was incorrectly invoking the unidentified logic when an endpoint had actually been identified, causing log messages to be output. ASTERISK-26349 #close Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f
2016-09-09res/res_pjsip: Add preferred_codec_only config to pjsip endpoint.Aaron An
This patch add config to pjsip by endpoint. ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec ; rather than advertising all joint codec capabilities. This ; limits the other side's codec choice to exactly what we prefer. ASTERISK-26317 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762
2016-09-09res_pjsip: Do not crash on ACKs from unknown endpoints.Mark Michelson
The endpoint identification PJSIP module is intended to identify which endpoint an incoming request is from. If an endpoint is not identified, then an artificial endpoint is used in its place when proceeding. The problem is that the ACK request type is an exception to the rule. The artificial endpoint is not used when processing an ACK. This results in the possibility of having a NULL endpoint being used further on. The reason ACK is an exception is an attempt not to spam security logs with unidentified requests. Presumably, you've already logged the unidentified request on the preceeding INVITE. Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion didn't cause an issue. A new change in 13.10 added endpoint ACL checking shortly after endpoint identification. Because we are accessing a NULL endpoint, this ACL check resulted in a crash. The fix here is to be sure to retrieve the artificial endpoint for all request types. ACKs still do not generate unidentified request security events. ASTERISK-26264 #close Reported by nappsoft AST-2016-006 Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703
2016-09-09chan_sip: Don't allocate new RTP instances on top of old ones.Joshua Colp
In some scenarios dialog_initialize_rtp can be called multiple times on the same dialog. This can cause RTP instances to be leaked along with multiple file descriptors for each instance. This change makes it so the existing RTP instances are destroyed and not overwritten, stopping the memory leak. ASTERISK-26272 #close patches: ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73
2016-09-08Merge "res_pjsip: Allow global headers to be overridden."zuul
2016-09-07Merge "ConfBridge: Make some announcements asynchronous."zuul
2016-09-07Merge "res/res_stasis_playback: Cancel the entire playlist when a stop occurs"zuul
2016-09-07Merge "apps/app_dial: Fix crash on non-connect call paths for ↵zuul
Privacy/Screening option"
2016-09-07res_pjsip: Allow global headers to be overridden.Joshua Colp
Currently when you add global headers from the dialplan both the header in the dialplan and the globally configured header are added to the resulting SIP INVITE. This change makes it so the headers in the dialplan take precedence and are the only ones added. Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad
2016-09-07Merge "res_resolver_unbound: Fix config documentation."zuul
2016-09-07Merge "res_pjsip_session: segfault on already disconnected session"zuul
2016-09-07Merge "apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5"zuul
2016-09-07ConfBridge: Make some announcements asynchronous.Mark Michelson
Confbridge announcements tend to block a channel while they are being played. In some circumstances, this is warranted since you want that particular channel not to hear the announcement (Example: "John Doe has entered the conference"). For others it makes less sense. This change first introduces methods for playing sounds asynchronously into the conference. This is very similar to how synchronous sounds are played, except the channel initiating the playback does not wait for the sound to complete before moving on. Asynchronous announcements are used for two circumstances: * Sounds played for a user after they have left the bridge * Sounds that play first to a single user and then the rest of the conference (if the channel and conference use the same language) ASTERISK-26289 #close Reported by Mark Michelson Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a
2016-09-07Merge "build: Add download capability for external packages"zuul
2016-09-07chan_sip: Allow Preferred sRTP.Alexander Traud
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
2016-09-07res_resolver_unbound: Fix config documentation.Joshua Colp
The code was referencing the config section as 'globals' instead of 'general'. This change swaps it over to 'general'. Change-Id: I9dfe7788f41c4a6754c77e103880dc1a747de7fe
2016-09-07Merge "chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP."Joshua Colp
2016-09-07Merge "pjsip_configuration.c: Ignore repeated identify by methods."Joshua Colp
2016-09-07Merge "resource_channels.c: add hangup reason "answered_elsewhere"."zuul
2016-09-06Merge "res_pjsip_registrar.c: Reduce stack usage in find_aor_name()."zuul
2016-09-06Merge "config_global.c: Comments and a default expression adjustment."zuul
2016-09-06Merge "sip_to_pjsip.py: Map canreinvite as directmedia alias."zuul
2016-09-06res/res_stasis_playback: Cancel the entire playlist when a stop occursMatt Jordan
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
2016-09-06Merge "sip_to_pjsip.py: Fix typo converting outboundproxy registration."zuul
2016-09-06Merge "sip_to_pjsip.py: Fix comment typo and tabs."zuul
2016-09-06Merge "Sample configs: Eliminate false multiline comment block starts."zuul
2016-09-06Merge "sorcery: Create function ast_sorcery_lockable_alloc."zuul
2016-09-06Merge "named_locks: Use ao2_weakproxy to deal with cleanup from container."zuul
2016-09-06build: Add download capability for external packagesGeorge Joseph
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
2016-09-06Merge "format_cap.c: Fix CLI "core show channeltype Surrogate" crash."Joshua Colp
2016-09-06Merge "astobj2: Support using a separate object for locking."zuul
2016-09-06res_pjsip_session: segfault on already disconnected sessionAlexei Gradinari
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
2016-09-06chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP.Walter Doekes
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
2016-09-04Merge "codecs: Add Codec 2 mode 2400."Joshua Colp
2016-09-04Merge "app_mp3: Use correct buffer size and the same sample rate as the channel"zuul
2016-09-03apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening optionMatt Jordan
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
2016-09-03apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5Matt Jordan
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
2016-09-02res_pjsip_registrar.c: Reduce stack usage in find_aor_name().Richard Mudgett
Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09
2016-09-02pjsip_configuration.c: Ignore repeated identify by methods.Richard Mudgett
Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838
2016-09-02config_global.c: Comments and a default expression adjustment.Richard Mudgett
Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3
2016-09-02sip_to_pjsip.py: Map canreinvite as directmedia alias.Richard Mudgett
Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2
2016-09-02sip_to_pjsip.py: Fix typo converting outboundproxy registration.Richard Mudgett
Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15
2016-09-02sip_to_pjsip.py: Fix comment typo and tabs.Richard Mudgett
Change-Id: If35174614545727817d329c60ba4456c028941b5
2016-09-02Sample configs: Eliminate false multiline comment block starts.Richard Mudgett
Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6
2016-09-02format_cap.c: Fix CLI "core show channeltype Surrogate" crash.Richard Mudgett
* Make ast_format_cap_get_names() NULL tolerant. ASTERISK-26331 #close Reported by: CGI.NET Change-Id: Id67e93936dc8ec2a33a9d33655843d43b59285a3
2016-09-02sorcery: Create function ast_sorcery_lockable_alloc.Corey Farrell
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
2016-09-02named_locks: Use ao2_weakproxy to deal with cleanup from container.Corey Farrell
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
2016-09-02astobj2: Support using a separate object for locking.Corey Farrell
Create ao2_alloc_with_lockobj function to support shared locking. Change-Id: Iba687eb9843922be7e481e23a32c0700ecf88a80
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