asterisk

Commit Graph

Author	SHA1	Message	Date
Jenkins2	3e8eea0325	Merge "res_pjsip: New endpoint option "refer_blind_progress"" into 13	8 years ago
Sean Bright	971a401ce9	sip.conf.sample: Clarify where DTLS settings are permitted ASTERISK-25101 #close Change-Id: I09a97793e5577b4422d0ae883fadb3f0d86725cc	8 years ago
Rodrigo Ramírez Norambuena	5da91c65be	Fix spelling queues.conf.sample file Change-Id: Ie1c2d83af66f27a449da09a68d987e0992627fee	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
Jenkins2	6383d9214a	Merge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages." into 13	8 years ago
Joshua Colp	10a49ab362	res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages. This change adds the required logic to allow the SIP Call-ID to be placed into the HEP RTCP traffic if the chan_sip module is used. In cases where the option is enabled but the channel is not either SIP or PJSIP then the code will fallback to the channel name as done previously. Based on the change on Nir's branch at: team/nirs/hep-chan-sip-support ASTERISK-26427 Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d	8 years ago
George Joseph	7d4a22bf2e	logger: Added logger_queue_limit to the configuration options. All log messages go to a queue serviced by a single thread which does all the IO. This setting controls how big that queue can get (and therefore how much memory is allocated) before new messages are discarded. The default is 1000. Should something go bezerk and log tons of messages in a tight loop, this will prevent memory escalation. When the limit is reached, a WARNING is logged to that effect and messages are discarded until the queue is empty again. At that time another WARNING will be logged with the count of discarded messages. There's no "low water mark" for this queue because the logger thread empties the entire queue and processes it in 1 batch before going back and waiting on the queue again. Implementing a low water mark would mean additional locking as the thread processes each message and it's not worth it. A "test" was added to test_logger.c but since the outcome is non-deterministic, it's really just a cli command, not a unit test. Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1	8 years ago
Richard Mudgett	cd80af508e	res_rtp_asterisk.c: Add stun_blacklist option Added the stun_blacklist option to rtp.conf. Some multihomed servers have IP interfaces that cannot reach the STUN server specified by stunaddr. Blacklist those interface subnets from trying to send a STUN packet to find the external IP address. Attempting to send the STUN packet needlessly delays processing incoming and outgoing SIP INVITEs because we will wait for a response that can never come until we give up on the response. Multiple subnets may be listed. ASTERISK-26890 #close Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342	8 years ago
Walter Doekes	bbbd262ec0	samples: Undo removal of include from canonicalize-app-names commit. This include was accidentally removed in changeset Ia79aea64de89531362e993e34230c2044a70aa93. My bad. Change-Id: I1d716c7f9590b4e97909fb8bca1f2ed9bd0e4082	8 years ago
zuul	b82f58d2bf	Merge "pjsip: Add Alembic for PUBLISH support." into 13	8 years ago
zuul	293f9d39b7	Merge "samples: Canonicalize app names in extensions.conf.sample." into 13	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
George Joseph	d2a33cdedc	sample_config: Add samples for pubsub to pjsip.conf.sample Added: * outbound-publish * resource_list * inbound-publication * asterisk-publication Change-Id: I65043a896c35483f30a92d30b5b118359af7ba5a	8 years ago
Walter Doekes	ab9d2fc86d	samples: Canonicalize app names in extensions.conf.sample. This takes care of warnings by ossobv/asterisklint. Change-Id: Ia79aea64de89531362e993e34230c2044a70aa93	8 years ago
George Joseph	0667febf3d	Merge "res_pjsip_config_wizard: Add 2 new parameters to help with proxy config" into 13	8 years ago
George Joseph	a827892ff7	res_pjsip_config_wizard: Add 2 new parameters to help with proxy config Two new parameters have been added to the pjsip config wizard. * Setting 'sends_line_with_registrations' to true will cause the wizard to skip the creation of an identify object to match incoming request to the endpoint and instead add the line and endpoint parameters to the outbound registration object. * Setting 'outbound_proxy' is a shortcut for adding individual endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy parameters. Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0	8 years ago
Sean Bright	2c28f7a922	res_musiconhold: Document the 'format' option ASTERISK-26086 #close Reported by: Jens Bürger Change-Id: I6aab666c0bf01fd0c64d7a5bcb22fa7f5d41335e	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
Sean Bright	8721d0bf1b	chan_sip: Add rtcp-mux support ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639	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
Matt Jordan	51985565ef	configs/samples/hep.conf.sample: Clarify how the HEP stack works This patch updates the documenation in hep.conf.sample to better specify how the various HEP modules interact. ASTERISK-26717 #close Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124	8 years ago
Daniel Journo	bc6eeab822	app_voicemail: Cannot set fromstring on a per-mailbox basis * apps/app_voicemail.c fromstring field added to mailbox which will override the global fromstring if set. ASTERISK-24562 #close Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe	8 years ago
zuul	59a00786e8	Merge "config: Improve documentation and behavior of outbound_proxy option." into 13	8 years ago
Tzafrir Cohen	5b34b751a0	pjsip.conf.sample: user_agent: not a specific version Use the description of useragent from sip.conf here. ASTERISK-26825 #close Change-Id: I5b33a4aaa0ae1d793289d05e3bc09521affbf755	8 years ago
Joshua Colp	d49af061bc	config: Improve documentation and behavior of outbound_proxy option. This change updates the documentation for the outbound_proxy option to ensure it is consistently stated that a full SIP URI must be provided for the option. The res_pjsip_outbound_registration module has also been changed so that the provided outbound_proxy value is checked to ensure it is a URI and if not an error is output stating so. ASTERISK-26782 Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593	8 years ago
Richard Mudgett	9f11da85a2	res_pjsip: Update authentication realm documentation. Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. An empty inbound auth realm represents the global section's default_realm value when the authentication object is used to challenge an incoming request. An empty outgoing auth realm is treated as a don't care wildcard when the authentication object is used to respond to an incoming authentication challenge. ASTERISK-26799 Change-Id: Id3952f7cfa1b6683b9954f2c5d2352d2f11059ce	8 years ago
Sean Bright	4c31e03e80	app_voicemail: Allow 'Comedian Mail' branding to be overriden Original patch by John Covert, slight modifications by me. ASTERISK-17428 #close Reported by: John Covert Patches: app_voicemail.c.patch (license #5512) patch uploaded by John Covert Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6	8 years ago
George Joseph	c7fcc4468f	configs/samples: Fix placement of 'identify' entry in sorcery.conf The entry for 'identify' was incorrectly placed in the res_pjsip section when it should be in res_pjsip_endpoint_identifier_ip. ASTERISK-26785 #close Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a	8 years ago
Sean Bright	70aff89e5d	res_odbc: Remove deprecated settings from sample configuration file ASTERISK-26704 #close Reported by: Anthony Messina Change-Id: I976a1f94cf79c5f31e76174c61f5c6a65fd6354f	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`)	9 years ago
George Joseph	47474cfd54	debug_utilities: Create the ast_coredumper utility This utility allows easy manipulation of asterisk coredumps. * Configurable search paths and patterns for existing coredumps * Can generate a consistent coredump from the running instance * Can dump the lock_infos table from a coredump * Dumps backtraces to separate files... - thread apply 1 bt full -> <coredump>.thread1.txt - thread apply all bt -> <coredump>.brief.txt - thread apply all bt full -> <coredump>.full.txt - lock_infos table -> <coredump>.locks.txt * Can tarball corefiles and optionally delete them after processing * Can tarball results files and optionally delete them after processing * Converts ':' in coredump and results file names '-' to facilitate uploading. Jira for instance, won't accept file names with colons in them. Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1]. [1] For *BSDs, the "devel/gdb" package might have to be installed to get a recent gdb. The utility will check all instances of gdb it finds in $PATH and if one isn't found that can run python, it prints a friendly error. Change-Id: I935d37ab9db85ef923f32b05579897f0893d33cd	9 years ago
Richard Mudgett	44fe4a5769	PJPROJECT logging: Made easier to get available logging levels. Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389	9 years ago
Alexander Traud	0cf1778eed	rtp_engine: Allow more than 32 dynamic payload types. The dynamic range (96-127) allows 32 RTP Payload Types. RFC 3551 section 3 allows to reassign other ranges. Consequently, when the dynamic range is exhausted, you can go for "rtp_pt_dynamic = 35" (or 0) in asterisk.conf. This enables the range 35-63 (or 0-63) giving room for another 29 (or 64) payload types. ASTERISK-26311 #close Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964 (cherry picked from commit `9ac53877f6`)	9 years ago
Kevin Harwell	50fa868ab8	codecs.conf.sample: Add sample and option descriptions for codec_opus codecs.conf.sample was missing codec opus's configuration options, descriptions, and examples. This patch adds the configuration options and examples to codecs.conf.sample that can be used with codec_opus. ASTERISK-26538 #close Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b	9 years ago
Rusty Newton	87903a6848	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	e0bc17edff	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	bb982480d8	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	3c62b60e56	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
Tzafrir Cohen	42cc267016	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
Joshua Colp	f16ab19292	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
Richard Mudgett	30af92e78d	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
Richard Mudgett	0f9b144c1a	Sample configs: Eliminate false multiline comment block starts. Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6	9 years ago
zuul	d4a4a23818	Merge "sip.conf: tlsclientmethod is using sslv23 as default." into 13	9 years ago
Alexander Traud	b1fe070d0b	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	329507fe20	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	050b90769a	Merge "core: Entity ID is not set or invalid" into 13	9 years ago
Alexei Gradinari	403c794684	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	2735ec899a	manager: Clarify that dialplan manipulation actions are under system class. ASTERISK-26246 #close Change-Id: Id673b9786389f9d2a87f638ce1a25161f5f31657	9 years ago
Alexei Gradinari	a06a1af0eb	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
Joshua Colp	bf21359292	Merge "app_voicemail: Add taskprocessor alert level options." into 13	9 years ago

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1862 Commits (812f5b51cb56a36668decc6dfc83adeca185429e)