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2017-08-15configure: Check cache for valid pjproject tarball before downloading.Richard Mudgett
On a fresh Asterisk source directory, the bundled pjproject tarball is unconditionally downloaded even if the tarball is already in a specified cache directory. * Made check if the pjproject tarball is valid in the cache directory before downloading the tarball on a fresh source directory. Change-Id: Ic7ec842d3c97ecd8dafbad6f056b7fdbce41cae5
2017-08-15res_xmpp: Google OAuth 2.0 protocol support for XMPP / MotifAndrey Egorov
Add ability to use tokens instead of passwords according to Google OAuth 2.0 protocol. ASTERISK-27169 Reported by: Andrey Egorov Tested by: Andrey Egorov Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db
2017-08-14Merge "STUN/netsock2: Fix some valgrind uninitialized memory findings."Jenkins2
2017-08-14Merge "res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown."George Joseph
2017-08-14Merge "res_pjsip: Remove ephemeral registered contacts on transport shutdown."George Joseph
2017-08-14Merge "res_pjsip: PJSIP Transport state monitor refactor."George Joseph
2017-08-14Merge "res_pjsip_transport_management.c: Rename some variables."Jenkins2
2017-08-10STUN/netsock2: Fix some valgrind uninitialized memory findings.Richard Mudgett
* netsock2.c: Test the addr->len member first as it may be the only member initialized in the struct. * stun.c:ast_stun_handle_packet(): The combinded[] local array could get used uninitialized by ast_stun_request(). The uninitialized string gets copied to another location and could overflow the destination memory buffer. These valgrind findings were found for ASTERISK_27150 but are not necessarily a fix for the issue. Change-Id: I55f8687ba4ffc0f69578fd850af006a56cbc9a57
2017-08-10res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown.Richard Mudgett
The fix for the issue is broken up into three parts. This is part three which handles the client side of REGISTER requests. The registered contact may no longer be valid on the server when the transport used is reliable and the connection is broken. * Re-REGISTER our contact if the reliable transport is broken after registration completes. We attempt to re-REGISTER immediately to minimize the time we are unreachable. Time may have already passed between the connection being broken and the loss being detected. * Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's are still correct if an allocation failure happens. ASTERISK-27147 Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83
2017-08-10res_pjsip: Remove ephemeral registered contacts on transport shutdown.Richard Mudgett
The fix for the issue is broken up into three parts. This is part two which handles the server side of REGISTER requests when rewrite_contact is enabled. Any registered reliable transport contact becomes invalid when the transport connection becomes disconnected. * Monitor the rewrite_contact's reliable transport REGISTER contact for shutdown. If it is shutdown then the contact must be removed because it is no longer valid. Otherwise, when the client attempts to re-REGISTER it may be blocked because the invalid contact is there. Also if we try to send a call to the endpoint using the invalid contact then the endpoint is not likely to see the request. The endpoint either won't be listening on that port for new connections or a NAT/firewall will block it. * Prune any rewrite_contact's registered reliable transport contacts on boot. The reliable transport no longer exists so the contact is invalid. * Websockets always rewrite the REGISTER contact address and the transport needs to be monitored for shutdown. * Made the websocket transport set a unique name since that is what we use as the ao2 container key. Otherwise, we would not know which transport we find when one of them shuts down. The names are also used for PJPROJECT debug logging. * Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state event. Now the global keep_alive_interval option, initially idle shutdown timer, and the server REGISTER contact monitor can work on wetsocket transports. * Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction. Now initially idle websockets will automatically shutdown. ASTERISK-27147 Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4
2017-08-10res_pjsip: PJSIP Transport state monitor refactor.Richard Mudgett
The fix for the issue is broken up into three parts. This is part one which refactors the transport state monitor code to allow more modules to be able to monitor transports. * Pull the management of PJPROJECT's transport state callback code from res_pjsip_transport_management.c into res_pjsip. Now other modules can dynamically add and remove themselves from transport monitoring without worrying about breaking PJPROJECT's callback chain. * Add the ability for other modules to get a callback whenever a specific transport is shutdown. ASTERISK-27147 Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912
2017-08-10res_pjsip_transport_management.c: Rename some variables.Richard Mudgett
* Use monitored instead of the misleading keepalive name. Change-Id: I9e5bcbb4ab2b82d49bcd0f06dfe85d15e0b552b6
2017-08-10UPGRADE notes: Prepare for the eventual 16 branch.Richard Mudgett
Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c
2017-08-10res_pjsip_messaging: IPv6 receive address needs bracketsScott Griepentrog
When handling an incoming SIP MESSAGE, PJSIP attaches the IP address that the message was received from to the message in the variable PJSIP_RECVADDR. When the IP address is IPv6 the :PORT appended results in an unparseable mess. By using an additional bit flag on the pj_sockaddr_print call, the conventional use of brackets around the address is achieved. ASTERISK-27193 #close Change-Id: I12342521f2ce87a5b6e4883d480a3fd957aa9fd9
2017-08-10Merge "Make --with-pjproject-bundled the default for Asterisk 15"Jenkins2
2017-08-09Merge "res_rtp_asterisk: Make P2P bridge Asymmetric codec aware"Jenkins2
2017-08-09res_rtp_asterisk: enable rtcp & QOS stats on native bridgeTorrey Searle
Asterisk wasn't generating or forwarding RTCP packets when native bridge was activated. Also the stats weren't available via CHANNEL(qos). Now the RTCP stats are always calculated. ASTERISK-27158 #close Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b
2017-08-09res_rtp_asterisk: Make P2P bridge Asymmetric codec awareTorrey Searle
Introduce a new property to rtp-engine to make it aware of the desire for assymetric codecs or not. If asymmetric codecs is not allowed, the bridge will compare read/write formats and shut down the p2p bridge if needed ASTERISK-26745 #close Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f
2017-08-09Merge "res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect"Jenkins2
2017-08-08Make --with-pjproject-bundled the default for Asterisk 15George Joseph
'--with-pjproject-bundled' is now the default when running ./configure. It can be disabled with '--without-pjproject-bundled'. To make building without an internet connection easier, a new ./configure option '--with-download-cache' was added that sets the cache for externals (like pjproject, the codecs and the DPMA), AND the sounds files. It can also be specified as an environment variable named "AST_DOWNLOAD_CACHE". The existing '--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and '--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable remain and if specified, will override '--with-downloads-cache'. ASTERISK-27189 Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce
2017-08-07res_pjsip_session: Release media resources on session end quicker.Joshua Colp
A change was made long ago where the session was kept around until the underlying INVITE session had been destroyed. This had the side effect of also keeping the underlying media resources around for this time as well. This change ensures that when we are told to terminate the session we immediately release any media sessions associated with it. ASTERISK-27110 Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82
2017-08-07Merge "bridge: Fix stream topology/participant locking and video misrouting."Jenkins2
2017-08-07Merge "chan_sip: Access incoming REFER headers in dialplan"Joshua Colp
2017-08-07Merge "channel: Fix leak on successful call to chan->tech->requester."Jenkins2
2017-08-07Merge "res_pjsip_nat.c: Remove unnecessary CMP_STOP."Joshua Colp
2017-08-07Merge "Support GMIME 3.0"Jenkins2
2017-08-07Merge "app_privacy: remove unused header asterisk/image.h"Jenkins2
2017-08-07chan_sip: Access incoming REFER headers in dialplankkm
This adds a way to access information passed along with SIP headers in a REFER message that initiates a transfer. Headers matching a dialplan variable GET_TRANSFERRER_DATA in the transferrer channel are added to a HASH object TRANSFER_DATA to be accessed with functions HASHKEY and HASH. The variable GET_TRANSFERRER_DATA is interpreted to be a prefix for headers that should be put into the hash. If not set, no headers are included. If set to a string (perhaps 'X-' in a typical case), all headers starting this string are added. Empty string matches all headers. If there are multiple of the same header, only the latest occurrence in the REFER message is available in the hash. Obviously, the variable GET_TRANSFERRER_DATA must be inherited by the referrer channel, and should be set with the '_' or '__' prefix. I avoided a specific reference to SIP or REFER, as in my mind the mechanism can be generalized to other channel techs. ASTERISK-27162 Change-Id: I73d7a1e95981693bc59aa0d5093c074b555f708e
2017-08-06bridge: Fix stream topology/participant locking and video misrouting.Joshua Colp
This change fixes a few locking issues and some video misrouting. 1. When accessing the stream topology of a channel the channel lock must be held to guarantee the topology remains valid. 2. When a channel was joined to a bridge the bridge specific implementation for stream mapping was not invoked, causing video to be misrouted for a brief period of time. ASTERISK-27182 Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03
2017-08-05channel: Fix leak on successful call to chan->tech->requester.Corey Farrell
joint_cap needs to be released unconditionally as chan->tech->requester does not steal the reference even on success. ASTERISK-27180 #close Change-Id: I647728992559bdb0a9c7357c20be1b36400d68b6
2017-08-04res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrectKevin Harwell
Currently, the handling of the msid attribute is not quite right. According to the spec the msid's between the offer/answer are not dependent upon one another. Meaning the same msid's given in an offer do not have to be returned in the answer for a given stream. And they probably shouldn't be (copied/reused) since this can potentially cause some browser side confusion. This patch generates new msids when both an offer and answer are sent from Asterisk. However, Asterisk does reuse the original msid it sent out for a reinvite. Also audio+video streams are paired together by sharing the same stream id, but a different track id. ASTERISK-27179 #close Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643
2017-08-04Merge "alembic/res_pjsip: Add "webrtc" configuration option"Jenkins2
2017-08-04Merge "chan_sip: Add dialplan function SIP_HEADERS"Joshua Colp
2017-08-04Merge "Fix compile error for old versions of GCC."Jenkins2
2017-08-04Merge "Correct some leaks in unit tests."Jenkins2
2017-08-04Merge "res_pjsip_transport_websocket.c: Fix serializer ref leak."Jenkins2
2017-08-04Merge "res_pjsip_outbound_registration.c: Misc fixes."Jenkins2
2017-08-03Correct some leaks in unit tests.Corey Farrell
* chan_sip: channel in test_sip_rtpqos_1. * test_config: config hook, config info and global config holder. * test_core_format: format in format_attribute_set_without_interface. * test_stream: unneeded frame duplication. * test_taskprocessor: task_data. Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31
2017-08-03res_pjsip_transport_websocket.c: Fix serializer ref leak.Richard Mudgett
Change-Id: Ib5a19bfd597f63d9021baeb645fc11153b3afa57
2017-08-03res_pjsip_outbound_registration.c: Misc fixes.Richard Mudgett
* Remove unnecessary CMP_STOP. * In handle_client_registration() use DEBUG_ATLEAST() to only do work needed for the debug log message when the debug log message is needed. * In sip_outbound_registration_state_destroy() check state->registration for NULL. Change-Id: I656d0fa11dda0b00048103efb1558e67a426fd80
2017-08-03res_pjsip_nat.c: Remove unnecessary CMP_STOP.Richard Mudgett
Change-Id: I6279b0d723bc3b75b8d65e81e02da9ea9bc0c3da
2017-08-03res_pjsip_registrar.c: Remove unnecessary CMP_STOP.Richard Mudgett
Most uses of CMP_STOP are superfluous and are only respected when OBJ_MULTIPLE is used to search the container. Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8
2017-08-03Support GMIME 3.0Tzafrir Cohen
Support building the Asterisk httpd with version 3.0 of gmime as well as earlier versions of that library. ASTERISK-27173 Change-Id: I7e13dd05a3083ccb0df2dabf83110223f6a9fa8f
2017-08-03alembic/res_pjsip: Add "webrtc" configuration optionKevin Harwell
When the "webrtc" option was added in res_pjsip it was not added to the alembic scripts. This patch adds the option for alembic. Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of an OPT_BOOL_T so if this field is ever written to a database it will write out the correct value. ASTERISK-27119 #close Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b
2017-08-02chan_sip: Add dialplan function SIP_HEADERSkkm
Syntax: SIP_HEADERS([prefix]) If the argument is specified, only the headers matching the given prefix are returned. The function returns a comma-separated list of SIP header names from an incoming INVITE message. Multiple headers with the same name are included in the list only once. The returned list can be iterated over using the functions POP() and SIP_HEADER(). For example, '${SIP_HEADERS(Co)}' might return the string 'Contact,Content-Length,Content-Type'. Practical use is rather '${SIP_HEADERS(X-)}' to enumerate optional extended headers sent by a peer. ASTERISK-27163 Change-Id: I2076d3893d03a2f82429f393b5b46db6cf68a267
2017-08-02Fix compile error for old versions of GCC.Corey Farrell
Use -Wno-format-truncation only if supported by compiler. ASTERISK-27171 #close Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6
2017-08-02app_privacy: remove unused header asterisk/image.hCorey Farrell
Change-Id: I56ed530633a642633b18383821069e806c92ae82
2017-08-01res_pjsip_pidf_eyebeam_body_supplement: Correct status presentationSean Bright
This change fixes PIDF content generation when the underlying device state is considered in use. Previously it was incorrectly marked as closed meaning they were offline/unavailable. The code now correctly marks them as open. Additionally: * Generate an XML element for our activity instead of a using a text node. * Consider every extension state other than "unavailable" to be 'open' status. * Update the XML namespaces and structure to reflect those documented in RFC 4480 * Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the "in use" activity. This change results in eyeBeam using the appropriate icon for the watched user. This was tested on eyeBeam 1.5.20.2 build 59030 on Windows. ASTERISK-26659 #close Reported by: Abraham Liebsch patches: ASTERISK-26659.diff submitted by snuffy (license 5024) Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810
2017-08-01res_pjsip: Add support for dnsmgr to external_media_address.Joshua Colp
The "external_media_address" option on transports is now resolved using dnsmgr. This allows it to be automatically refreshed regularly if refreshes are enabled in dnsmgr. If the system is using a dynamic IP address a dynamic DNS hostname can be provided to keep the IP address up to date. Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2
2017-08-01Fix compiler warnings on Fedora 26 / GCC 7.Corey Farrell
GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
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