1856 Commits (273debd261148dcc0962566d83ee9c601ee5c99c)

Author	SHA1	Message	Date
Rusty Newton	badd38f031	SAC documentation: don't specify transports for endpoints and registrations ... Removing explicit transport definition for endpoints and registrations. It isn't necessary and isn't generally advised. ASTERISK-26514 #close Change-Id: Ifdec5e631962438a4683600968dfa4bfd15909fb	9 years ago
Joshua Colp	aed6c219a3	pjsip: Fix a few media bugs with reinvites and asymmetric payloads. ... When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc	9 years ago
Joshua Colp	403c4f5833	pjsip: Support dual stack automatically. ... This change adds support for dual stack automatically. No configuration is required and the IP address and version in the SIP messages and SDP will be automatically changed based on the transport over which the message is being sent. RTP usage has also been changed to listen on both IPv4 and IPv6 simultaneously to allow media to flow, and to allow ICE support on both simultaneously. This also allows failover between IPv6 and IPv4 to work as expected. ASTERISK-26309 #close Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d	9 years ago
Michael Walton	3e96d491d0	res_rtp_asterisk: Add ice_blacklist option ... Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the form ice_blacklist = <subnet spec>, e.g. ice_blacklist = 192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay discovery. This is useful for optimizing the ICE process where a system has multiple host address ranges and/or physical interfaces and certain of them are not expected to be used for RTP. Multiple ice_blacklist configuration lines may be used. If left unconfigured, all discovered host addresses are used, as per previous behavior. Documention in rtp.conf.sample. ASTERISK-26418 #close Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9	9 years ago
Ludovic Gasc (GMLudo)	9f62feca60	res_calendar: Add support for fetching calendars when reloading ... We use a lot res_calendar, we are very happy with that, especially because you use libical, the almost alone opensource library that supports really ical format with all types of recurrency. Nevertheless, some features are missed for our business use cases. This first patch adds a new option in calendar.conf: fetch_again_at_reload. Be my guest for a better name. If it's true, when you'll launch "module reload res_calendar.so", Asterisk will download again the calendar. The business use case is that we have a WebUI with a scheduler planner, we know when the calendars are modified. For now, we need to define 1 minute of timeout to have a chance that our user doesn't wait too long between the modification and the real test. But it generates a lot of useless HTTP traffic. ASTERISK-26422 #close Change-Id: I384b02ebfa42b142bbbd5b7221458c7f4dee7077	9 years ago
Joshua Colp	8966c8ec5a	Merge "cdr_mysql: fix UTC support"	9 years ago
Joshua Colp	78b6190a11	odbc: Remove options that are no longer applicable. ... The pooling, shared_connection, limit, and idlecheck options are no longer used in res_odbc. ASTERISK-26389 Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6	9 years ago
Tzafrir Cohen	d3ddf4b0fd	cdr_mysql: fix UTC support ... * Make 'cdrzone=UTC' work properly. * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone ASTERISK-26359 #close Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778	9 years ago
Richard Mudgett	ba362822f3	res_pjsip: Add ignore_uri_user_options option. ... 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	9 years ago
zuul	9d54dd04bb	Merge "res/res_pjsip: Add preferred_codec_only config to pjsip endpoint."	9 years ago
Aaron An	2a50c29101	res/res_pjsip: Add preferred_codec_only config to pjsip endpoint. ... 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	9 years ago
Richard Mudgett	4aaa27e532	Sample configs: Eliminate false multiline comment block starts. ... Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6	9 years ago
zuul	1c64616373	Merge "sip.conf: tlsclientmethod is using sslv23 as default."	9 years ago
Alexander Traud	1a9555f036	sip.conf: tlsclientmethod is using sslv23 as default. ... When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL SSLv23_method. This was documented incorrectly in the file sip.conf.sample. SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that function should have been called 'secure_method' or 'automatic_method' back in the 90s. Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if you face a server which has problems like not falling back to TLSv1.0 automatically. ASTERISK-24425 Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3	9 years ago
George Joseph	534063fd67	res_pjsip: Add contact_user to endpoint ... contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4	9 years ago
zuul	9fc83f8ffd	Merge "core: Entity ID is not set or invalid"	9 years ago
Alexei Gradinari	e85adbd947	core: Entity ID is not set or invalid ... The Exchanging Device and Mailbox States could not working if the Entity ID (EID) is not set manually and can't be obtained from ethernet interface. This patch replaces debug message to warning and addes missing description about option 'entityid' to asterisk.conf.sample. With this patch the asterisk also: (1) decline loading the modules which won't work without EID: res_corosync and res_pjsip_publish_asterisk. (2) warn if EID is empty on loading next modules: pbx_dundi, res_xmpp Starting with v197 systemd/udev will automatically assign "predictable" names for all local Ethernet interfaces. This patch also addes some new ethernet prefixes "eno" and "ens". ASTERISK-26164 #close Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6	9 years ago
Joshua Colp	922b74169f	manager: Clarify that dialplan manipulation actions are under system class. ... ASTERISK-26246 #close Change-Id: Id673b9786389f9d2a87f638ce1a25161f5f31657	9 years ago
George Joseph	36b2a40533	autohints: Update CHANGES and extensions.conf.sample ... Make it clear that we're talking about device state hints and add an entry to the sample config. Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433	9 years ago
Alexei Gradinari	403b63571c	res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack ... The PJSIP taskprocessors could be overflowed on startup if there are many (thousands) realtime endpoints configured with unsolicited mwi. The PJSIP stack could be totally unresponsive for a few minutes after boot completed. This patch creates a separate PJSIP serializers pool for mwi and makes unsolicited mwi use serializers from this pool. This patch also adds 2 new global options to tune taskprocessor alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'. This patch also adds new global option 'mwi_disable_initial_unsolicited' to disable sending unsolicited mwi to all endpoints on startup. If disabled then unsolicited mwi will start processing on next endpoint's contact update. ASTERISK-26230 #close Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a	9 years ago
Alexei Gradinari	9042ad40f2	app_voicemail: Add taskprocessor alert level options. ... On heavy loaded system with IMAP or DB storage, 'app_voicemail' taskprocessor queue could reach 500 scheduled tasks. It could happen when the IMAP or DB server dies or is unreachable. It could happen on startup when there are many (thousands) realtime endpoints configured with unsolicited mwi. If the taskprocessor queue reaches the high water level then the alert is triggered and pjsip stops processing new requests until the queue reaches the low water level to clear the alert. This patch adds 2 new 'general' configuration options to tune taskprocessor alert levels: 'tps_queue_high' - Taskprocessor high water alert trigger level. 'tps_queue_low' - Taskprocessor low water clear alert level ASTERISK-26229 #close Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8	9 years ago
Richard Mudgett	327136088e	dsp.c: Correct DTMF twist dsp.conf documentation. ... Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae	9 years ago
Richard Mudgett	4286a369a1	res_pjsip: Whitespace and comment cleanup. ... Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38	9 years ago
Richard Mudgett	0d1744e132	chan_dahdi: Add faxdetect_timeout option. ... The new option allows the channel driver's faxdetect option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. * Don't clear dsp_features after passing them to the dsp code in my_pri_ss7_open_media(). We should still remember them especially for the new faxdetect_timeout option. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Ieffd3fe788788d56282844774365546dce8ac810	9 years ago
Richard Mudgett	e739888d99	res_pjsip: Add fax_detect_timeout endpoint option. ... The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d	9 years ago
Matt Jordan	dab2a6b689	hep.conf.sample: Default 'enabled' to 'no' ... Following the principle of least surprise, we should not be sending massive numbers of PJSIP and RTCP HEP packets out into the ether to some only-slightly-random IP address. Having 'enabled' set to 'no' in the sample configuration file should prevent this from happening for those who run 'make samples'. ASTERISK-26159 #close Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1	9 years ago
Matt Jordan	83f2c2573b	configs/basic-pbx/modules.conf: Remove 'bad' modules ... This patch removes the following modules: - pbx_functions: It never existed. - res_pjsip_log_forwarder: It no longer exists. - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs aren't going to be installing HOMER - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't loaded, and we aren't configured to make use of the module Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5	9 years ago
zuul	88dfcd21b2	Merge "chan_sip: Support auth username for callbackextension feature"	10 years ago
Joshua Colp	31a5c28339	res_odbc: Implement a connection pool. ... Testing has shown that our usage of UnixODBC is problematic due to bugs within UnixODBC itself as well as the heavy weight cost of connecting and disconnecting database connections, even when pooling is enabled. For users of UnixODBC 2.3.1 and earlier crashes would occur due to insufficient protection of the disconnect operation. This was fixed in UnixODBC 2.3.2 and above. For users of UnixODBC 2.3.3 and higher a slow-down would occur under heavy database use due to repeated connection establishment. A regression is present where on each connection the database configuration is cached again, with the cache growing out of control. The connection pool implementation present in this change helps to mitigate these issues by reducing how much we connect and disconnect database connections. We also solve the issue of crashes under UnixODBC 2.3.1 by defaulting the maximum number of connections to 1, returning us to the previous working behavior. For users who may have a fixed version the maximum concurrent connection limit can be increased helping with performance. The connection pool works by keeping a list of active connections. If the connection limit has not been reached a new connection is established. If the connection limit has been reached then the request waits until a connection becomes available before continuing. ASTERISK-26074 #close ASTERISK-26054 #close Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff	10 years ago
Timo Teräs	538c6415c6	chan_sip: Support auth username for callbackextension feature ... ASTERISK-20527 #close Change-Id: I659cf7f00836a09d09d146ad226a40477d731239	10 years ago
Tzafrir Cohen	1d60bfcdf1	followme: allow disabling callee prompt ... Add the option 'enable_callee_prompt' to followme.conf. Enabled by default. If disabled, a callee is not prompted to accept or reject the forwarded call. ASTERISK-26064 #close Change-Id: I0a8b19d4cf95c86a07c992813babb9e4a4acfff5 Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	10 years ago
Joshua Colp	56f24345f7	Merge "func_odbc: single database connection should be optional"	10 years ago
Alexei Gradinari	c378b00a83	func_odbc: single database connection should be optional ... func_odbc was changed in Asterisk 13.9.0 to make func_odbc use a single database connection per DSN because of reported bug ASTERISK-25938 with MySQL/MariaDB LAST_INSERT_ID(). This is drawback in performance when func_odbc is used very often in dialplan. Single database connection should be optional. ASTERISK-26010 Change-Id: I7091783a7150252de8eeb455115bd00514dfe843	10 years ago
Joshua Colp	b57032c364	Merge "res_hep: Provide an option to pick the UUID type"	10 years ago
Matt Jordan	a1803cb5f4	configs/samples/pjsip.conf.sample: Fix typo ... A ':' is not a valid token for starting a comment. Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad	10 years ago
Matt Jordan	e06a23681c	res_hep: Provide an option to pick the UUID type ... At one point in time, it seemed like a good idea to use the Asterisk channel name as the HEP correlation UUID. In particular, it felt like this would be a useful identifier to tie PJSIP messages and RTCP messages together, along with whatever other data we may eventually send to Homer. This also had the benefit of keeping the correlation UUID channel technology agnostic. In practice, it isn't as useful as hoped, for two reasons: 1) The first INVITE request received doesn't have a channel. As a result, there is always an 'odd message out', leading it to be potentially uncorrelated in Homer. 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. This causes RTCP information to be uncorrelated to the SIP message traffic seen by those capture nodes. In order to support both (in case someone is trying to use res_hep_rtcp with a non-PJSIP channel), this patch adds a new option, uuid_type, with two valid values - 'call-id' and 'channel'. The uuid_type option is used by a module to determine the preferred UUID type. When available, that source of a correlation UUID is used; when not, the more readily available source is used. For res_hep_pjsip: - uuid_type = call-id: the module uses the SIP Call-ID header value - uuid_type = channel: the module uses the channel name if available, falling back to SIP Call-ID if not For res_hep_rtcp: - uuid_type = call-id: the module uses the SIP Call-ID header if the channel type is PJSIP and we have a channel, falling back to the Stasis event provided channel name if not - uuid_type = channel: the module uses the channel name ASTERISK-25352 #close Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c	10 years ago
Joshua Colp	1bfe8602a5	Merge "basic-cfg: asterisk.conf: don't set languages"	10 years ago
Joshua Colp	da2506b2dc	Merge "basic-cfg: asterisk.conf: debug level 5 spams"	10 years ago
Joshua Colp	d733dccf81	Merge "basic-cfg: asterisk.conf: defaults of options"	10 years ago
zuul	c5cb9d120f	Merge "basic-cfg: asterisk.conf: remove [directories]"	10 years ago
Tzafrir Cohen	ec85ea3c21	basic-cfg: asterisk.conf: don't set languages ... * No need to set language in a miniml configuration. 'en' will do just fine. * It would be useful to have an example of setting it to a different language. * Setting the documentation language explicitly is likewise not required. Setting it to a different value is not common. At least until there is a set of translated documentation. Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7 Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	10 years ago
Tzafrir Cohen	1b0a9bb2c4	basic-cfg: asterisk.conf: debug level 5 spams ... Don't suggest users to use debug level 5, which spews (usually non-useful) debug information. Reduce the suggestion to (an arbitrarily-selected) level 2. Change-Id: Ib53195f78945970956ff59ef13fa89b90e0fcd60 Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	10 years ago
Tzafrir Cohen	d0ba3e8196	basic-cfg: asterisk.conf: defaults of options ... Note the default of remmed-out options. To clarify that those values are not the defaults. Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738 Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	10 years ago
Tzafrir Cohen	f943a1fd84	basic-cfg: asterisk.conf: remove [directories] ... A minimal configuration does not need to explicitly spell out the directories. The built-in defaults will do just fine. In many cases they are wrong. Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	10 years ago
Jaco Kroon	8923c9ac96	app_confbridge: Add a regcontext option for confbridge bridge profiles. ... This patch allows for having app_confbridge register the name of the conference as an extension into a specific context, similar to regcontext for chan_sip. This variant is not quite as involved as the one in chan_sip and doesn't allow for multiple contexts or custom extensions, you can only specify the context and the conference name will always be used as the extension to register. ASTERISK-25989 #close Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f	10 years ago
Diederik de Groot	bf13b59062	configs/basic-pbx/asterisk.conf: contains incorrect path separator ... Note: When packagers use these files (as an example) the paths are never really used when they are split using '='. Note: Thirdparty applications will also have trouble parsing the file when expecting '=>'. Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004	10 years ago
George Joseph	4ebf9a938d	res_pjsip: Add ability to identify by Authorization username ... A feature of chan_sip that service providers relied upon was the ability to identify by the Authorization username. This is most often used when customers have a PBX that needs to register rather than identify by IP address. From my own experiance, this is pretty common with small businesses who otherwise don't need a static IP. In this scenario, a register from the customer's PBX may succeed because From will usually contain the PBXs account id but an INVITE will contain the caller id. With nothing recognizable in From, the service provider's Asterisk can never match to an endpoint and the INVITE just stays unauthorized. The fixes: A new value "auth_username" has been added to endpoint/identify_by that will use the username and digest fields in the Authorization header instead of username and domain in the the From header to match an endpoint, or the To header to match an aor. This code as added to res_pjsip_endpoint_identifier_user rather than creating a new module. Although identify_by was always a comma-separated list, there was only 1 choice so order wasn't preserved. So to keep the order, a vector was added to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar to find the aor. The res_pjsip_endpoint_identifier_* modules are called in globals/endpoint_identifier_order. Along the way, the logic in res_pjsip_registrar was corrected to match most-specific to least-specific as res_pjsip_endpoint_identifier_user does. The order is: username@domain username@domain_alias username Auth by username does present 1 problem however, the first INVITE won't have an Authorization header so the distributor, not finding a match on anything, sends a securty_alert. It still sends a 401 with a challenge so the next INVITE will have the Authorization header and presumably succeed. As a result though, that first security alert is actually a false alarm. To address this, a new feature has been added to pjsip_distributor that keeps track of unidentified requests and only sends the security alert if a configurable number of unidentified requests come from the same IP in a configurable amout of time. Those configuration options have been added to the global config object. This feature is only used when auth_username is enabled. Finally, default_realm was added to the globals object to replace the hard coded "asterisk" used when an endpoint is not yet identified. The testsuite tests all pass but new tests are forthcoming for this new feature. ASTERISK-25835 #close Reported-by: Ross Beer Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d	10 years ago
Alexei Gradinari	860b135c88	res_pjsip: disable multi domain to improve realtime performace ... This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7	10 years ago
Leif Madsen	6ede210c98	Remove reference to non-existent sip.conf option ... Option was removed in commit 7f883ef495 ASTERISK-25927 #close Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8	10 years ago
George Joseph	e2524fcee3	res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited ... res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes on endpoints for unsolicited mwi and on aors for subscriptions required that the admin know in advance which the client wanted. If you specified mailboxes on the endpoint, subscriptions were rejected even if you also specified mailboxes on the aor. Voicemail extension: * Added a global default_voicemail_extension which defaults to "". * Added voicemail_extension to both endpoint and aor. * Added ast_sip_subscription_get_dialog for support. * Added ast_sip_subscription_get_sip_uri for support. When an unsolicited NOTIFY is constructed, the From header is parsed, the voicemail extension from the endpoint is substituted for the user, and the result placed in the Message-Account field in the body. When a subscribed NOTIFY is constructed, the subscription dialog local uri is parsed, the voicemail_extension from the aor (looked up from the subscription resource name) is substituted for the user, and the result placed in the Message-Account field in the body. If no voicemail extension was defined, the Message-Account field is not added to the NOTIFY body. mwi_subscribe_replaces_unsolicited: * Added mwi_subscribe_replaces_unsolicited to endpoint. The previous behavior was to reject a subscribe if a previous internal subscription for unsolicited MWI was found for the mailbox. That remains the default. However, if there are mailboxes also set on the aor and the client subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal subscription is removed and replaced with the external subscription. This allows an admin to configure mailboxes on both the endpoint and aor and allows the client to select which to use. ASTERISK-25865 #close Reported-by: Ross Beer Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea	10 years ago