429 Commits (22.0.0-pre1)

Author	SHA1	Message	Date
Mike Bradeen	b2455b2732	res_pjsip_notify: add dialplan application ... Add dialplan application PJSIPNOTIFY to send either pre-configured NOTIFY messages from pjsip_notify.conf or with headers defined in dialplan. Also adds the ability to send pre-configured NOTIFY commands to a channel via the CLI. Resolves: #799 UserNote: A new dialplan application PJSIPNotify is now available which can send SIP NOTIFY requests from the dialplan. The pjsip send notify CLI command has also been enhanced to allow sending NOTIFY messages to a specific channel. Syntax: pjsip send notify <option> channel <channel>	1 year ago
Ben Ford	fb0e46e464	channel: Add multi-tenant identifier. ... This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID.	1 year ago
gibbz00	99a5064a07	feat: ARI "ChannelToneDetected" event ... A stasis event is now produced when using the TONE_DETECT dialplan function. This event is published over ARI using the ChannelToneDetected event. This change does not make it available over AMI. Fixes: #811 UserNote: Setting the TONE_DETECT dialplan function on a channel in ARI will now cause a ChannelToneDetected ARI event to be raised when the specified tone is detected.	1 year ago
George Joseph	a02fc685a8	stir_shaken: CRL fixes and a new CLI command ... * Fixed a bug in crypto_show_cli_store that was causing asterisk to crash if there were certificate revocation lists in the verification certificate store. We're also now prefixing certificates with "Cert:" and CRLs with "CRL:" to distinguish them in the list. * Added 'untrusted_cert_file' and 'untrusted_cert_path' options to both verification and profile objects. If you have CRLs that are signed by a different CA than the incoming X5U certificate (indirect CRL), you'll need to provide the certificate of the CRL signer here. Thse will show up as 'Untrusted" when showing the verification or profile objects. * Fixed loading of crl_path. The OpenSSL API we were using to load CRLs won't actually load them from a directory, only a file. We now scan the directory ourselves and load the files one-by-one. * Fixed the verification flags being set on the certificate store. - Removed the CRL_CHECK_ALL flag as this was causing all certificates to be checked for CRL extensions and failing to verify the cert if there was none. This basically caused all certs to fail when a CRL was provided via crl_file or crl_path. - Added the EXTENDED_CRL_SUPPORT flag as it is required to handle indirect CRLs. * Added a new CLI command... `stir_shaken verify certificate_file <certificate_file> [ <profile> ]` which will assist troubleshooting certificate problems by allowing the user to manually verify a certificate file against either the global verification certificate store or the store for a specific profile. * Updated the XML documentation and the sample config file. Resolves: #809	1 year ago
George Joseph	79e0b50162	voicemail.conf.sample: Fix ':' comment typo ... ...and removed an errant trailing space. Resolves: #819	1 year ago
George Joseph	6e73af341a	app_voicemail_odbc: Allow audio to be kept on disk ... This commit adds a new voicemail.conf option 'odbc_audio_on_disk' which when set causes the ODBC variant of app_voicemail to leave the message and greeting audio files on disk and only store the message metadata in the database. This option came from a concern that the database could grow to large and cause remote access and/or replication to become slow. In a clustering situation with this option, all asterisk instances would share the same database for the metadata and either use a shared filesystem or other filesystem replication service much more suitable for synchronizing files. The changes to app_voicemail to implement this feature were actually quite small but due to the complexity of the module, the actual source code changes were greater. They fall into the following categories: * Tracing. The module is so complex that it was impossible to figure out the path taken for various scenarios without the addition of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in code that's not related to the functional change. Making this worse was the fact that many "if" statements in this module didn't use braces. Since the tracing macros add multiple statements, many "if" statements had to be converted to use braces. * Excessive use of PATH_MAX. Previous maintainers of this module used PATH_MAX to allocate character arrays for filesystem paths and SQL statements as though they cost nothing. In fact, PATH_MAX is defined as 4096 bytes! Some functions had (and still have) multiples of these. One function has 7. Given that the vast majority of installations use the default spool directory path `/var/spool/asterisk/voicemail`, the actual path length is usually less than 80 bytes. That's over 4000 bytes wasted. It was the same for SQL statement buffers. A 4K buffer for statement that only needed 60 bytes. All of these PATH_MAX allocations in the ODBC related code were changed to dynamically allocated buffers. The rest will have to be addressed separately. * Bug fixes. During the development of this feature, several pre-existing ODBC related bugs were discovered and fixed. They had to do with leaving orphaned files on disk, not preserving original message ids when moving messages between folders, not honoring the "formats" config parameter in certain circumstances, etc. UserNote: This commit adds a new voicemail.conf option 'odbc_audio_on_disk' which when set causes the ODBC variant of app_voicemail_odbc to leave the message and greeting audio files on disk and only store the message metadata in the database. Much more information can be found in the voicemail.conf.sample file.	1 year ago
Alexei Gradinari	07aa0ffe6a	app_queue: Add option to not log Restricted Caller ID to queue_log ... Add a queue option log-restricted-caller-id to strip the Caller ID when storing the ENTERQUEUE event in the queue log if the Caller ID is restricted. Resolves: #765 UpgradeNote: Add a new column to the queues table: queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes') to control whether the Restricted Caller ID will be stored in the queue log. UserNote: Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID will be stored in the queue log. If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.	1 year ago
George Joseph	85241bd229	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." ... This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	1 year ago
Ivan Poddubny	abe0018d85	configs: Fix a misleading IPv6 ACL example in Named ACLs ... "deny=::" is equivalent to "::/128". In order to mean "deny everything by default" it must be "::/0".	2 years ago
George Joseph	758ed2b9fd	stir_shaken: Fix memory leak, typo in config, tn canonicalization ... * Fixed possible memory leak in tn_config:tn_get_etn() where we weren't releasing etn if tn or eprofile were null. * We now canonicalize TNs before using them for lookups or adding them to Identity headers. * Fixed a typo in stir_shaken.conf.sample. Resolves: #716	2 years ago
Sperl Viktor	0ab4a5ef6b	res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address. ... Add a new identify_by option to res_pjsip_endpoint_identifier_ip called 'transport' this matches endpoints based on the bound ip address (local) instead of the 'ip' option, which matches on the source ip address (remote). UserNote: set identify_by=transport for the pjsip endpoint. Then use the existing 'match' option and the new 'transport' option of the identify. Fixes: #672	2 years ago
Sperl Viktor	895ab9d798	res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI ... Add ability to match against PJSIP request URI. UserNote: this new feature let users match endpoints based on the indound SIP requests' URI. To do so, add 'request_uri' to the endpoint's 'identify_by' option. The 'match_request_uri' option of the identify can be an exact match for the entire request uri, or a regular expression (between slashes). It's quite similar to the header identifer. Fixes: #599	2 years ago
Joshua Elson	c8ab570c6f	Implement Configurable TCP Keepalive Settings in PJSIP Transports ... This commit introduces configurable TCP keepalive settings for both TCP and TLS transports. The changes allow for finer control over TCP connection keepalives, enhancing stability and reliability in environments prone to connection timeouts or where intermediate devices may prematurely close idle connections. This has proven necessary and has already been tested in production in several specialized environments where access to the underlying transport is unreliable in ways invisible to the operating system directly, so these keepalive and timeout mechanisms are necessary. Fixes #657	2 years ago
Naveen Albert	b5850941b1	app_voicemail: Allow preventing mark messages as urgent. ... This adds an option to allow preventing callers from leaving messages marked as 'urgent'. Resolves: #619 UserNote: The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'.	2 years ago
George Joseph	628f8d7a43	Stir/Shaken Refactor ... Why do we need a refactor? The original stir/shaken implementation was started over 3 years ago when little was understood about practical implementation. The result was an implementation that wouldn't actually interoperate with any other stir-shaken implementations. There were also a number of stir-shaken features and RFC requirements that were never implemented such as TNAuthList certificate validation, sending Reason headers in SIP responses when verification failed but we wished to continue the call, and the ability to send Media Key(mky) grants in the Identity header when the call involved DTLS. Finally, there were some performance concerns around outgoing calls and selection of the correct certificate and private key. The configuration was keyed by an arbitrary name which meant that for every outgoing call, we had to scan the entire list of configured TNs to find the correct cert to use. With only a few TNs configured, this wasn't an issue but if you have a thousand, it could be. What's changed? * Configuration objects have been refactored to be clearer about their uses and to fix issues. * The "general" object was renamed to "verification" since it contains parameters specific to the incoming verification process. It also never handled ca_path and crl_path correctly. * A new "attestation" object was added that controls the outgoing attestation process. It sets default certificates, keys, etc. * The "certificate" object was renamed to "tn" and had it's key change to telephone number since outgoing call attestation needs to look up certificates by telephone number. * The "profile" object had more parameters added to it that can override default parameters specified in the "attestation" and "verification" objects. * The "store" object was removed altogther as it was never implemented. * We now use libjwt to create outgoing Identity headers and to parse and validate signatures on incoming Identiy headers. Our previous custom implementation was much of the source of the interoperability issues. * General code cleanup and refactor. * Moved things to better places. * Separated some of the complex functions to smaller ones. * Using context objects rather than passing tons of parameters in function calls. * Removed some complexity and unneeded encapsuation from the config objects. Resolves: #351 Resolves: #46 UserNote: Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information. UpgradeNote: The stir-shaken refactor is a breaking change but since it's not working now we don't think it matters. The stir_shaken.conf file has changed significantly which means that existing ones WILL need to be changed. The stir_shaken.conf.sample file in configs/samples/ has quite a bit more information. This is also an ABI breaking change since some of the existing objects needed to be changed or removed, and new ones added. Additionally, if res_stir_shaken is enabled in menuselect, you'll need to either have the development package for libjwt v1.15.3 installed or use the --with-libjwt-bundled option with ./configure.	2 years ago
Mike Bradeen	39760d109b	res_pjsip: disable raw bad packet logging ... Add patch to split the log level for invalid packets received on the signaling port. The warning regarding the packet will move to level 2 so that it can still be displayed, while the raw packet will be at level 4.	2 years ago
Naveen Albert	d1fb397cfc	general: Fix broken links. ... This fixes a number of broken links throughout the tree, mostly caused by wiki.asterisk.org being replaced with docs.asterisk.org, which should eliminate the need for sporadic fixes as in f28047db36. Resolves: #430	2 years ago
Naveen Albert	44b955e237	configs: Improve documentation for bandwidth in iax.conf. ... This improves the documentation for the bandwidth setting in iax.conf by making it clearer what the ramifications of this setting are. It also changes the sample default from low to high, since only high is compatible with good codecs that people will want to use in the vast majority of cases, and this is a common gotcha that trips up new users. Resolves: #425	2 years ago
Sean Bright	c040179fcf	doc: Update IP Quality of Service links. ... Fixes #328	2 years ago
Samuel Olaechea	3287585379	configs: Fix typo in pjsip.conf.sample.	2 years ago
Naveen Albert	37e4af6a09	chan_dahdi: Clarify scope of callgroup/pickupgroup. ... Internally, chan_dahdi only applies callgroup and pickupgroup to FXO signalled channels, but this is not documented anywhere. This is now documented in the sample config, and a warning is emitted if a user tries configuring these settings for channel types that do not support these settings, since they will not have any effect. Resolves: #294	2 years ago
George Joseph	67fd66e2fe	file.c: Add ability to search custom dir for sounds ... To better co-exist with sounds files that may be managed by packages, custom sound files may now be placed in AST_DATA_DIR/sounds/custom instead of the standard AST_DATA_DIR/sounds/<lang> directory. If the new "sounds_search_custom_dir" option in asterisk.conf is set to "true", asterisk will search the custom directory for sounds files before searching the standard directory. For performance reasons, the "sounds_search_custom_dir" defaults to "false". Resolves: #315 UserNote: A new option "sounds_search_custom_dir" has been added to asterisk.conf that allows asterisk to search AST_DATA_DIR/sounds/custom for sounds files before searching the standard AST_DATA_DIR/sounds/<lang> directory.	2 years ago
Jaco Kroon	4db98a38f1	app_queue: periodic announcement configurable start time. ... This newly introduced periodic-announce-startdelay makes it possible to configure the initial start delay of the first periodic announcement after which periodic-announce-frequency takes over. UserNote: Introduce a new queue configuration option called 'periodic-announce-startdelay' which will vary the normal (historic) behavior of starting the periodic announcement cycle at periodic-announce-frequency seconds after entering the queue to start the periodic announcement cycle at period-announce-startdelay seconds after joining the queue. The default behavior if this config option is not set remains unchanged. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	2 years ago
Naveen Albert	301b0258bf	sig_analog: Add Called Subscriber Held capability. ... This adds support for Called Subscriber Held for FXS lines, which allows users to go on hook when receiving a call and resume the call later from another phone on the same line, without disconnecting the call. This is a convenience mechanism that most real PSTN telephone switches support. ASTERISK-30372 #close Resolves: #240 UserNote: Called Subscriber Held is now supported for analog FXS channels, using the calledsubscriberheld option. This allows a station user to go on hook when receiving an incoming call and resume from another phone on the same line by going on hook, without disconnecting the call.	2 years ago
Sean Bright	3b806a3303	extconfig: Allow explicit DB result set ordering to be disabled. ... Added a new boolean configuration flag - `order_multi_row_results_by_initial_column` - to both res_pgsql.conf and res_config_odbc.conf that allows the administrator to disable the explicit `ORDER BY` that was previously being added to all generated SQL statements that returned multiple rows. Fixes: #179	2 years ago
Naveen Albert	d16046e41f	chan_dahdi: Allow autoreoriginating after hangup. ... Currently, if an FXS channel is still off hook when all calls on the line have hung up, the user is provided reorder tone until going back on hook again. In addition to not reflecting what most commercial switches actually do, it's very common for switches to automatically reoriginate for the user so that dial tone is provided without the user having to depress and release the hookswitch manually. This can increase convenience for users. This behavior is now supported for kewlstart FXS channels. It's supported only for kewlstart (FXOKS) mainly because the behavior doesn't make any sense for ground start channels, and loop start signalling doesn't provide the necessary DAHDI event that makes this easy to implement. Likely almost everyone is using FXOKS over FXOLS anyways since FXOLS is pretty useless these days. ASTERISK-30357 #close Resolves: #224 UserNote: The autoreoriginate setting now allows for kewlstart FXS channels to automatically reoriginate and provide dial tone to the user again after all calls on the line have cleared. This saves users from having to manually hang up and pick up the receiver again before making another call.	2 years ago
Naveen Albert	e1a1ae933b	sig_analog: Allow three-way flash to time out to silence. ... sig_analog allows users to flash and use the three-way dial tone as a primitive hold function, simply by never timing it out. Some systems allow this dial tone to time out to silence, so the user is not annoyed by a persistent dial tone. This option allows the dial tone to time out normally to silence. ASTERISK-30004 #close Resolves: #205 UserNote: The threewaysilenthold option now allows the three-way dial tone to time out to silence, rather than continuing forever.	2 years ago
Sean Bright	5c4cbeff87	extensions.conf.sample: Remove reference to missing context. ... c3ff4648 removed the [iaxtel700] context but neglected to remove references to it. This commit addresses that and also removes iaxtel and freeworlddialup references from other config files.	2 years ago
Naveen Albert	dd171a44b7	users.conf: Deprecate users.conf configuration. ... This deprecates the users.conf config file, which is no longer as widely supported but still integrated with a number of different modules. Because there is no real mechanism for marking a configuration file as "deprecated", and users.conf is not just used in a single place, this now emits a warning to the user when the PBX loads to notify about the deprecation. This configuration mechanism has been widely criticized and discouraged since its inception, and is no longer relevant to the configuration that most users are doing today. Removing it will allow for some simplification and cleanup in the codebase. Resolves: #183 UpgradeNote: The users.conf config is now deprecated and will be removed in a future version of Asterisk.	2 years ago
Naveen Albert	8cd7548e43	sig_analog: Allow immediate fake ring to be suppressed. ... When immediate=yes on an FXS channel, sig_analog will start fake audible ringback that continues until the channel is answered. Even if it answers immediately, the ringback is still audible for a brief moment. This can be disruptive and unwanted behavior. This adds an option to disable this behavior, though the default behavior remains unchanged. ASTERISK-30003 #close Resolves: #118 UserNote: The immediatering option can now be set to no to suppress the fake audible ringback provided when immediate=yes on FXS channels.	2 years ago
Nathan Bruning	292834d1ba	app_queue: Add force_longest_waiting_caller option. ... This adds an option 'force_longest_waiting_caller' which changes the global behavior of the queue engine to prevent queue callers from 'jumping ahead' when an agent is in multiple queues. Resolves: #108 Also closes old asterisk issues: - ASTERISK-17732 - ASTERISK-17570 Change-Id: I0f84e27903fefbe2018d0afa2d67b23aa0b321ce	2 years ago
Naveen Albert	ce7a72d7e2	res_musiconhold: Add option to loop last file. ... Adds the loop_last option to res_musiconhold, which allows the last audio file in the directory to be looped perpetually once reached, rather than circling back to the beginning again. Resolves: #122 ASTERISK-30462 UserNote: The loop_last option in musiconhold.conf now allows the last file in the directory to be looped once reached.	2 years ago
Naveen Albert	273ad73d99	sig_analog: Add fuller Caller ID support. ... A previous change, ASTERISK_29991, made it possible to send additional Caller ID parameters that were not previously supported. This change adds support for analog DAHDI channels to now be able to receive these parameters for on-hook Caller ID, in order to enhance the usability of CPE that support these parameters. Resolves: #94 ASTERISK-30331 UserNote: Additional Caller ID properties are now supported on incoming calls to FXS stations, namely the redirecting reason and call qualifier.	2 years ago
Sean Bright	0d6b271831	core: Cleanup gerrit and JIRA references. (#58) ... * Remove .gitreview and switch to pulling the main asterisk branch version from configure.ac instead. * Replace references to JIRA with GitHub. * Other minor cleanup found along the way. Resolves: #39	3 years ago
Naveen Albert	9a999242b2	chan_dahdi: Add dialmode option for FXS lines. ... Currently, both pulse and tone dialing are always enabled on all FXS lines, with no way of disabling one or the other. In some circumstances, it is desirable or necessary to disable one of these, and this behavior can be problematic. A new "dialmode" option is added which allows setting the methods to support on a per channel basis for FXS (FXO signalled lines). The four options are "both", "pulse", "dtmf"/"tone", and "none". Additionally, integration with the CHANNEL function is added so that this setting can be updated for a channel during a call. Resolves: #35 ASTERISK-29992 UserNote: A "dialmode" option has been added which allows specifying, on a per-channel basis, what methods of subscriber dialing (pulse and/or tone) are permitted. Additionally, this can be changed on a channel at any point during a call using the CHANNEL function.	3 years ago
Joshua Colp	4ec4543332	Revert "app_queue: periodic announcement configurable start time." ... This reverts commit 71e317f68f. Reason for revert: Causes segmentation fault. Change-Id: I3beeda83249bffec2a8f246aa50a6b2f1b59ef59	3 years ago
Naveen Albert	ecf49ff746	pbx_dundi: Add PJSIP support. ... Adds PJSIP as a supported technology to DUNDi. To facilitate this, we now allow an endpoint to be specified for outgoing PJSIP calls. We also allow users to force a specific channel technology for outgoing SIP-protocol calls. ASTERISK-28109 #close ASTERISK-28233 #close Change-Id: I2e28e5a5d007bd49e3df113ad567b011103899bf	3 years ago
Naveen Albert	eadf28a476	voicemail.conf: Fix incorrect comment about #include. ... A comment at the top of voicemail.conf says that #include cannot be used in voicemail.conf because this breaks the ability for app_voicemail to auto-update passwords. This is factually incorrect, since Asterisk has no problem updating files that are #include'd in the main configuration file, and this does work in voicemail.conf as well. ASTERISK-30479 #close Change-Id: I3bf7d275849ab83f55f7fb6702a75a3077ee1df3	3 years ago
Jaco Kroon	71e317f68f	app_queue: periodic announcement configurable start time. ... This newly introduced periodic-announce-startdelay makes it possible to configure the initial start delay of the first periodic announcement after which periodic-announce-frequency takes over. ASTERISK-30437 #close Change-Id: Ia79984b6377ef78f167ad9ea2ac084bec29955d0 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	3 years ago
Naveen Albert	907692abb3	app_osplookup: Remove obsolete sample config. ... ASTERISK_30302 previously removed app_osplookup, but its sample config was not removed. This removes it since nothing else uses it. ASTERISK-30438 #close Change-Id: Ife234208f5f197644475db4ab1af95a8551642fd	3 years ago
Holger Hans Peter Freyther	92849c8c62	res_http_media_cache: Introduce options and customize ... Make the existing CURL parameters configurable and allow to specify the usable protocols, proxy and DNS timeout. ASTERISK-30340 Change-Id: I2eb02ef44190e026716720419bcbdbcc8125777b	3 years ago
Sean Bright	632da7c11a	app_queue: Reset all queue defaults before reload. ... Several queue fields were not being set to their default value during a reload. Additionally added some sample configuration options that were missing from queues.conf.sample. Change-Id: I3a88c7877af91752b1b46a0c087384f7eb9c47e4	3 years ago
Naveen Albert	d1bec3623e	res_pjsip_session: Add overlap_context option. ... Adds the overlap_context option, which can be used to explicitly specify a context to use for overlap dialing extension matches, rather than forcibly using the context configured for the endpoint. ASTERISK-30262 #close Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db	3 years ago
Mike Bradeen	6b03d60c7d	res_monitor: Remove deprecated module. ... ASTERISK-30303 Change-Id: I0462caefb4f9544e2e2baa23c498858310b52d50	3 years ago
Mike Bradeen	e8f548c155	app_macro: Remove deprecated module. ... For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6	3 years ago
Mike Bradeen	4095a382da	chan_sip: Remove deprecated module. ... ASTERISK-30297 Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996	3 years ago
Mike Bradeen	de3ce178ab	chan_alsa: Remove deprecated module. ... ASTERISK-30298 Change-Id: I5c8afb781528afdf55d237e3bffa5e4a862ae8c7	3 years ago
Mike Bradeen	89a7d30a97	chan_mgcp: Remove deprecated module. ... Also removes res_pktcops to avoid merge conflicts with ASTERISK~30301. ASTERISK-30299 Change-Id: I41a316d327646a197b6f112f7f637aceb5111b41	3 years ago
Michael Kuron	841107f294	res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip ... chan_sip supported sending AOC-D and AOC-E information in SIP INFO messages in an "AOC" header in a format that was originally defined by Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC format that is supported by devices from multiple vendors, including Snom phones with firmware >= 8.4.2 (released in 2010). This commit adds a new res_pjsip_aoc module that inserts AOC information into outgoing messages or sends SIP INFO messages as described below. It also fixes a small issue in res_pjsip_session which didn't always call session supplements on outgoing_response. * AOC-S in the 180/183/200 responses to an INVITE request * AOC-S in SIP INFO (if a 200 response has already been sent or if the INVITE was sent by Asterisk) * AOC-D in SIP INFO * AOC-D in the 200 response to a BYE request (if the client hangs up) * AOC-D in a BYE request (if Asterisk hangs up) * AOC-E in the 200 response to a BYE request (if the client hangs up) * AOC-E in a BYE request (if Asterisk hangs up) The specification defines one more, AOC-S in an INVITE request, which is not implemented here because it is not currently possible in Asterisk to have AOC data ready at this point in call setup. Once specifying AOC-S via the dialplan or passing it through from another SIP channel's INVITE is possible, that might be added. The SIP INFO requests are sent out immediately when the AOC indication is received. The others are inserted into an appropriate outgoing message whenever that is ready to be sent. In the latter case, the XML is stored in a channel variable at the time the AOC indication is received. Depending on where the AOC indications are coming from (e.g. PRI or AMI), it may not always be possible to guarantee that the AOC-E is available in time for the BYE. Successfully tested AOC-D and both variants of AOC-E with a Snom D735 running firmware 10.1.127.10. It does not appear to properly support AOC-S however, so that could only be tested by inspecting SIP traces. ASTERISK-21502 #close Reported-by: Matt Jordan <mjordan@digium.com> Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333	3 years ago
Naveen Albert	1c5738771d	res_hep: Add support for named capture agents. ... Adds support for the capture agent name field of the Homer protocol to Asterisk by allowing users to specify a name that will be sent to the HEP server. ASTERISK-30322 #close Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b	3 years ago