asterisk

Commit Graph

Author	SHA1	Message	Date
Jenkins2	0d105398f0	Merge "install_prereq: Update RHEL/CentOS/Fedora libraries." into 13	7 years ago
Alexander Traud	8daed642c7	install_prereq: Update RHEL/CentOS/Fedora libraries. deleted automake git ncurses-devel pjproject-devel sqlite2-devel libsqlite3x-devel renamed radiusclient-ng-devel to radcli-devel gmime22-devel to gmime-dev added alsa-lib-devel bash binutils-devel bison doxygen flex hoard make pkgconfig speexdsp-devel uriparser-devel uw-imap-devel wget xmlstarlet zlib-devel ASTERISK-27599 Reported by: Said Masoud Change-Id: I05bb0af98ae532b2d5f37478e38b8f0762b1c035	7 years ago
Alexander Traud	7ad6792011	install_prereq: Download latest Jansson. ASTERISK-27603 Change-Id: I65c587534c0ae364f063d68da1bed40bb3d5e8aa	7 years ago
Jenkins2	ede9185249	Merge "install_prereq: For PJProject, point users to configure script." into 13	7 years ago
Jenkins2	24def35abc	Merge "install_prereq: Support package manager DNF and yum option strict=1." into 13	7 years ago
Alexander Traud	d5a24cfdf7	install_prereq: For PJProject, point users to configure script. The installation script and the new configure option --with-pjproject-bundled aimed to accomplish the same. However, the installation script was out of date. Users should go for the maintained configure option, or the Wiki. ASTERISK-24598 Change-Id: Icbf4b562f81f7c05bd24a3805bd46c0beb4ebd44	7 years ago
Alexander Traud	f20a522d43	install_prereq: Update Debian/Ubuntu libraries. ASTERISK-27555 Change-Id: Ieb41b0cbf968af12882b39454b819ebb48b9ea46	7 years ago
Alexander Traud	1fbca9a40d	install_prereq: Support package manager DNF and yum option strict=1. This re-enables the script ./contrib/scripts/install_prereq on Fedora 22 and newer, and on RHEL/CentOS when the option strict=1 was set for yum install. ASTERISK-27598 Reported by: Hunter Stevens, Said Masoud Change-Id: I40f9517122aaa6906e8fc0962b4b8008dfddb368	7 years ago
Jenkins2	16f211c647	Merge "res_pjsip: Split type=identify to IP address and SIP header matching priorities" into 13	7 years ago
Alexander Traud	1d625a7428	install_prereq: Update Debian/Ubuntu libraries. ASTERISK-27555 Change-Id: I0818b6e42631be1b69237e2b41d3415275693e53	7 years ago
Richard Mudgett	f35960d55b	res_pjsip: Split type=identify to IP address and SIP header matching priorities The type=identify endpoint identification method can match by IP address and by SIP header. However, the SIP header matching has limited usefulness because you cannot specify the SIP header matching priority relative to the IP address matching. All the matching happens at the same priority and the order of evaluating the identify sections is indeterminate. e.g., If you had two type=identify sections where one matches by IP address for endpoint alice and the other matches by SIP header for endpoint bob then you couldn't predict which endpoint is matched when a request comes in that matches both. * Extract the SIP header matching criteria into its own "header" endpoint identification method so the user can specify the relative priority of the SIP header and the IP address matching criteria in the global endpoint_identifier_order option. The "ip" endpoint identification method now only matches by IP address. ASTERISK-27491 Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095	7 years ago
Sean Bright	ce3d56920b	Remove as much trailing whitespace as possible. Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Sean Bright	b4ae112e3a	Fix some invalid Unicode characters configs/samples/minivm.conf.sample contains invalid UTF-8, but that appears to be intentional. Change-Id: I7b1e0d332f3380fd0425962a3c9c55f9b200c8cc	8 years ago
George Joseph	062a4390ac	ast_coredumper: Add ability to use directory other than /tmp The OUTPUTDIR environment variable can now be set either in the environment itself or in ast_debug_tools.conf. If set, it's used for all work products instead of /tmp. Also added the --tarball-config option that includes the contents of /etc/asterisk when either --tarball-coredumps or --tarball-results are used. Change-Id: I66b2553319df61caea5b313d084f51978f730b4c	8 years ago
Alexander Traud	5f4197735d	install_prereq: Checkout of libSRTP 2.x. Since Asterisk 13.17, libSRTP 2.x is supported. Therefore, its latest version is installed again via the script install_prereq. ASTERISK-27356 Change-Id: I13125839a79052356469e41edacbebff0a937d39	8 years ago
Jenkins2	ae2687494a	Merge "ast_coredumper: allow setting asterisk binary explicitly" into 13	8 years ago
Jenkins2	835db751e1	Merge "ast_coredumper: Add gzipping of binaries and display of signal info" into 13	8 years ago
Tzafrir Cohen	1b535d8457	ast_coredumper: allow setting asterisk binary explicitly Adds an extra option, --asterisk-bin=<path> to ast_coredumper. If provided, the binary given to gdb will be the parameter, rather than asterisk from the PATH. ASTERISK-27380 #close Change-Id: I25f5b91eb75059b0fb2f142e468c26b283b0a9f3	8 years ago
Joshua Colp	7385d1e017	res_pjsip: Add 'ip' as a valid option to 'identify_by' on endpoint. When the identify_by option on an endpoint is set to ip it will only be identified using the res_pjsip_endpoint_identifier_ip module. This ensures that it is not mistakenly matched using the username of the From header. To ensure behavior has not changed the default has been changed to "username,ip" for the identify_by option. ASTERISK-27206 Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd	8 years ago
George Joseph	3f7ad66245	ast_coredumper: Add gzipping of binaries and display of signal info The --tarball-coredump option now creates a gzipped tarball of coredumps processed, their results txt files and copies of /etc/os-release, /usr/sbin/asterisk, /usr/lib(64)/libasterisk* and /usr/lib(64)/asterisk as those files are needed to properly examine the coredump. The file will be named /tmp/asterisk.<timestamp>.coredumps.tar.gz or /tmp/asterisk-<uniqueid>.coredumps.tar.gz if --tarball-uniqueid was specified. Added dumps of *_siginfo to the top of the txt files so you can tell what signal was invoked. Change-Id: Ib9ee6d83592d4b1bc90cb3419a05376a88d1ded9	8 years ago
Torrey Searle	407b78504d	contrib/script/sip_to_pjsip: implement 'all' for allow/disallow when 'all' is specified in an allow or disallow section, it should erase all values from the inverse section in the default config. E.G. allow=all should erase any deny values from default config & vice-versa ASTERISK-27333 #close Change-Id: I99219478fb98f08751d769daaee0b7795118a5a6	8 years ago
Torrey Searle	6c30f4a2d1	contrib/thirdparty/sip_to_pjsip: add additional flag mappings add mappings for udptl redundancy, rtptimeout, and debug flags Change-Id: Ie73cf5c83c05dee01eb9624ede76c1a30225d73a	8 years ago
Jenkins2	df7211421e	Merge "res_pjsip: Add handling for incoming unsolicited MWI NOTIFY" into 13	8 years ago
George Joseph	ed2a4ee81e	res_pjsip: Add handling for incoming unsolicited MWI NOTIFY A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c	8 years ago
Jenkins2	1aef4fe9df	Merge "alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode" into 13	8 years ago
George Joseph	5ff2d06aa6	alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode The downgrade function was missing "_v2" at the end of the alter column type. Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e	8 years ago
Florian Floimair	ad606844be	alembic: Add support for MS-SQL MS-SQL has no native Enum-type support and therefore needs to work with constraints. Since these constraints need unique names the suggested approach referenced in the following alembic documentation has been applied: http://bit.ly/2x9r8pb ASTERISK-27255 #close Change-Id: I4a399ba3eed41a33ce8cb294968ad340221580ee	8 years ago
Joshua Colp	9a366d2424	Merge "alembic: Fix enum creation for dtls_fingerprint" into 13	8 years ago
George Joseph	15ddc9acb3	alembic: Fix enum creation for dtls_fingerprint Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db	8 years ago
Florian Floimair	2370469645	alembic: fix erroneous commit for add_prune_on_boot Added include for postgresql ENUM type and redefined values in the same way as in the other migration scripts. ASTERISK-27254 #close Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa	8 years ago
Florian Floimair	bf178a0f4f	alembic: Add dtls_fingerprint column in ps_endpoints table The ps_endpoints table was missing the dtls_fingerprint column introduced with commit `adba2a8d7f`. ASTERISK-27168 #close Change-Id: I9cb5006f7f50718b5239919562773adabb334cfd	8 years ago
Richard Mudgett	07d026b4cd	res_pjsip: Remove ephemeral registered contacts on transport shutdown. 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	8 years ago
Sean Bright	cea4ce246d	corosync: Fix corosync library name in configure.ac Also add new corosync packages to install_prereq. Reported by Travis Ryan in #asterisk-dev Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db	8 years ago
George Joseph	379fe65831	Fix alembic branches Change-Id: I04f607f084bda9b1b7f626e8e9735c37dc751187	8 years ago
Torrey Searle	9fbc34d2bd	res_pjsip: Add DTMF INFO Failback mode The existing auto dtmf mode reverts to inband if 4733 fails to be negotiated. This patch adds a new mode auto_info which will switch to INFO instead of inband if 4733 is not available. ASTERISK-27066 #close Change-Id: Id185b11e84afd9191a2f269e8443019047765e91	8 years ago
Alexei Gradinari	a6e4899612	res_pjsip: New endpoint option "notify_early_inuse_ringing" This option was added to control whether to notify dialog-info state 'early' or 'confirmed' on Ringing when already INUSE. The value "yes" is useful for some SIP phones (Cisco SPA) to be able to indicate and pick up ringing devices. ASTERISK-26919 #close Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711	8 years ago
Alexei Gradinari	6af2dd34af	res_pjsip: New endpoint option "refer_blind_progress" This option was added to turn off notifying the progress details on Blind Transfer. If this option is not set then the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted". Some SIP phones like Mitel/Aastra or Snom keep the line busy until receive "200 OK". ASTERISK-26333 #close Change-Id: Id606fbff2e02e967c02138457badc399144720f2	8 years ago
Joshua Colp	c09b9dba90	alembic: Add table for 'resource_list' PJSIP RLS type. This change adds an Alembic migration which adds a ps_resource_list table that can contain resource_list RLS configuration objects. ASTERISK-26929 Change-Id: I7c888fafc67b3e87012de974f71ca7a5b8b1ec05	8 years ago
Joshua Colp	b3f4a6365e	pjsip: Add Alembic for PUBLISH support. This change adds database tables for the PUBLISH support so it can be configured using realtime. A minor fix to the res_pjsip_publish_asterisk module was done so that it read the sorcery configuration from the correct section. Finally the sample configuration files have been updated. ASTERISK-26928 Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952	8 years ago
Joshua Colp	1d1309b1ed	Revert "Update for 13.15.0-rc1" This reverts commit `552cf009c0`. Change-Id: Ie345bea481261b761c44079e9472622040fda302	8 years ago
Joshua Colp	e52ced02d8	Merge "cel_pgsql.c: Fix buffer overflow calling libpq" into 13	8 years ago
Sean Bright	864dda07f3	alembic: Turn off execute bit on non-executable python scripts Change-Id: I744c986da4a38aeff8c00837eb89de7841fbc86c	8 years ago
Kevin Harwell	552cf009c0	Update for 13.15.0-rc1	8 years ago
Richard Begg	398e5ec16c	res_pjsip_session: Enable RFC3578 overlap dialing support. Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6	8 years ago
George Joseph	9b756662a8	res_pjsip: Symmetric transports A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f	8 years ago
Mark Michelson	7bc69753bc	Add rtcp-mux support This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5	8 years ago
Matt Jordan	b3c2c996f1	res_pjsip_endpoint_identifier_ip: Add an option to match requests by header This patch adds a new features to the endpoint identifier module, 'match_header'. When set, inbound requests are matched by a provided SIP header: value pair. This option works in conjunction with the existing 'match' configuration option, such that if any 'match*' attribute matches an inbound request, the request is associated with the specified endpoint. Since this module now identifies by more than just IP address, appropriate renaming of the module and/or variables can be done in a non-release branch. ASTERISK-26863 #close Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453 (cherry picked from commit `30f52d79d7`)	8 years ago
George Joseph	7fd28cefdb	debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts Forgot to install it with the original patch Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c	8 years ago
George Joseph	456bc3c704	debug_utilities: Add ast_logescalator The escalator works by creating a set of startup commands in cli.conf that set up logger channels and issue the debug commands for the subsystems specified. If asterisk is running when it is executed, the same commands will be issued to the running instance. The original cli.conf is saved before any changes are made and can be restored by executing '$prog --reset'. The log output will be stored in... $astlogdir/message.$uniqueid $astlogdir/debug.$uniqueid $astlogdir/dtmf.$uniqueid $astlogdir/fax.$uniqueid $astlogdir/security.$uniqueid $astlogdir/pjsip_history.$uniqueid $astlogdir/sip_history.$uniqueid Some minor tweaks were made to chan_sip, and res_pjsip_history so their history output could be send to a log channel as packets are captured. A minor tweak was also made to manager so events are output to verbose when "manager set debug on" is issued. Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543	8 years ago
George Joseph	c628a7acac	debug_utilities: Create ast_loggrabber ast_loggrabber gathers log files from customizable search patterns, optionally converts POSIX timestamps to a readable format and tarballs the results. Also a few tweaks were made to ast_coredumper. Change-Id: I8bfe1468ada24c1344ce4abab7b002a59a659495 (cherry picked from commit `5fa1c56d7e`)	8 years ago

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615 Commits (c9bff6be9b65723e1cc6c42baec8d9045d0b80ef)