mirror of https://github.com/asterisk/asterisk
Asterisk Development Team
10 months ago
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19bbcf0bbf
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ChangeLogs/ChangeLog-20.8.1.md
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ChangeLogs/ChangeLog-20.9.0-rc1.md
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## Change Log for Release asterisk-20.9.0-rc1
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### Links:
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- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0-rc1.md)
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- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0-rc1)
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- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0-rc1.tar.gz)
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- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
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### Summary:
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- Commits: 20
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- Commit Authors: 9
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- Issues Resolved: 8
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- Security Advisories Resolved: 0
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### User Notes:
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- #### app_voicemail_odbc: Allow audio to be kept on disk
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This commit adds a new voicemail.conf option
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'odbc_audio_on_disk' which when set causes the ODBC variant of
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app_voicemail_odbc to leave the message and greeting audio files
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on disk and only store the message metadata in the database.
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Much more information can be found in the voicemail.conf.sample
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file.
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- #### app_queue: Add option to not log Restricted Caller ID to queue_log
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Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID
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will be stored in the queue log.
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If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.
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- #### pbx.c: expand fields width of "core show hints"
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The fields width of "core show hints" were increased.
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The width of "extension" field to 30 characters and
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the width of the "device state id" field to 60 characters.
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- #### rtp_engine: add support for multirate RFC2833 digits
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No change in configuration is required in order to enable this
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feature. Endpoints configured to use RFC2833 will automatically have this
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enabled. If the endpoint does not support this, it should not include it in
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the SDP offer/response.
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Resolves: #699
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### Upgrade Notes:
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- #### app_queue: Add option to not log Restricted Caller ID to queue_log
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Add a new column to the queues table:
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queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes')
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to control whether the Restricted Caller ID will be stored in the queue log.
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### Commit Authors:
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- Alexei Gradinari: (2)
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- Bastian Triller: (1)
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- Chrsmj: (1)
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- George Joseph: (4)
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- Igor Goncharovsky: (1)
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- Mike Bradeen: (2)
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- Sean Bright: (7)
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- Tinet-Mucw: (1)
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- Walter Doekes: (1)
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## Issue and Commit Detail:
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### Closed Issues:
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- 699: [improvement]: Add support for multi-rate DTMF
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- 736: [bug]: Seg fault on CLI after PostgreSQL CDR module fails to load for a second time
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- 765: [improvement]: Add option to not log Restricted Caller ID to queue_log
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- 770: [improvement]: pbx.c: expand fields width of "core show hints"
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- 776: [bug] DTMF broken after rtp_engine: add support for multirate RFC2833 digits commit
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- 783: [bug]: Under certain circumstances a channel snapshot can get orphaned in the cache
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- 789: [bug]: Mediasec headers aren't sent on outgoing INVITEs
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- 797: [bug]:
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### Commits By Author:
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- ### Alexei Gradinari (2):
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- pbx.c: expand fields width of "core show hints"
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- app_queue: Add option to not log Restricted Caller ID to queue_log
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- ### Bastian Triller (1):
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- cli: Show configured cache dir
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- ### George Joseph (4):
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- app_voicemail_odbc: Allow audio to be kept on disk
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- stasis_channels: Use uniqueid and name to delete old snapshots
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- security_agreement.c: Always add the Require and Proxy-Require headers
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- ast-db-manage: Remove duplicate enum creation
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- ### Igor Goncharovsky (1):
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- res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS()
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- ### Mike Bradeen (2):
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- rtp_engine: add support for multirate RFC2833 digits
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- res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits
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- ### Sean Bright (7):
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- file.h: Rename function argument to avoid C++ keyword clash.
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- bundled_pjproject: Disable UPnP support.
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- asterisk.c: Don't log an error if .asterisk_history does not exist.
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- xml.c: Update deprecated libxml2 API usage.
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- manager.c: Properly terminate `CoreShowChannelMap` event.
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- pjsip: Add PJSIP_PARSE_URI_FROM dialplan function.
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- logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.
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- ### Tinet-mucw (1):
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- bridge_basic.c: Make sure that ast_bridge_channel is not destroyed while itera..
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- ### Walter Doekes (1):
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- chan_ooh323: Fix R/0 typo in docs
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- ### chrsmj (1):
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- cdr_pgsql: Fix crash when the module fails to load multiple times.
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### Commit List:
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- res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS()
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- res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits
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- ast-db-manage: Remove duplicate enum creation
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- security_agreement.c: Always add the Require and Proxy-Require headers
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- logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.
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- stasis_channels: Use uniqueid and name to delete old snapshots
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- app_voicemail_odbc: Allow audio to be kept on disk
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- app_queue: Add option to not log Restricted Caller ID to queue_log
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- pbx.c: expand fields width of "core show hints"
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- pjsip: Add PJSIP_PARSE_URI_FROM dialplan function.
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- manager.c: Properly terminate `CoreShowChannelMap` event.
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- cli: Show configured cache dir
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- xml.c: Update deprecated libxml2 API usage.
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- cdr_pgsql: Fix crash when the module fails to load multiple times.
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- asterisk.c: Don't log an error if .asterisk_history does not exist.
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- chan_ooh323: Fix R/0 typo in docs
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- bundled_pjproject: Disable UPnP support.
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- file.h: Rename function argument to avoid C++ keyword clash.
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- rtp_engine: add support for multirate RFC2833 digits
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### Commit Details:
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#### res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS()
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Author: Igor Goncharovsky
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Date: 2024-05-12
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When using the PJSIP_DIAL_CONTACTS() function for use in the Dial()
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command, the contacts are returned in text form, so the input to
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the path_outgoing_request() function is a contact value of NULL.
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The issue was reported in ASTERISK-28211, but was not actually fixed
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in ASTERISK-30100. This fix brings back the code that was previously
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removed and adds code to search for a contact to extract the path
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value from it.
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#### res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits
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Author: Mike Bradeen
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Date: 2024-06-21
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After change made in 624f509 to add support for non 8K RFC 4733/2833 digits,
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Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of
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the negotiated codec(s).
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This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC
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offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec.
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A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to
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be re-written to allow for these scenarios.
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Fixes: #776
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#### ast-db-manage: Remove duplicate enum creation
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Author: George Joseph
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Date: 2024-07-08
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Remove duplicate creation of ast_bool_values from
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2b7c507d7d12_add_queue_log_option_log_restricted_.py. This was
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causing alembic upgrades to fail since the enum was already created
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in fe6592859b85_fix_mwi_subscribe_replaces_.py back in 2018.
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Resolves: #797
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#### security_agreement.c: Always add the Require and Proxy-Require headers
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Author: George Joseph
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Date: 2024-07-03
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The `Require: mediasec` and `Proxy-Require: mediasec` headers need
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to be sent whenever we send `Security-Client` or `Security-Verify`
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headers but the logic to do that was only in add_security_headers()
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in res_pjsip_outbound_register. So while we were sending them on
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REGISTER requests, we weren't sending them on INVITE requests.
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This commit moves the logic to send the two headers out of
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res_pjsip_outbound_register:add_security_headers() and into
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security_agreement:ast_sip_add_security_headers(). This way
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they're always sent when we send `Security-Client` or
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`Security-Verify`.
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Resolves: #789
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#### logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.
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Author: Sean Bright
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Date: 2024-06-29
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Fixes #785
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#### stasis_channels: Use uniqueid and name to delete old snapshots
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Author: George Joseph
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Date: 2024-05-08
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Whenver a new channel snapshot is created or when a channel is
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destroyed, we need to delete any existing channel snapshot from
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the snapshot cache. Historically, we used the channel->snapshot
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pointer to delete any existing snapshots but this has two issues.
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First, if something (possibly ast_channel_internal_swap_snapshots)
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sets channel->snapshot to NULL while there's still a snapshot in
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the cache, we wouldn't be able to delete it and it would be orphaned
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when the channel is destroyed. Since we use the cache to list
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channels from the CLI, AMI and ARI, it would appear as though the
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channel was still there when it wasn't.
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Second, since there are actually two caches, one indexed by the
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channel's uniqueid, and another indexed by the channel's name,
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deleting from the caches by pointer requires a sequential search of
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all of the hash table buckets in BOTH caches to find the matching
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snapshots. Not very efficient.
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So, we now delete from the caches using the channel's uniqueid
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and name. This solves both issues.
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This doesn't address how channel->snapshot might have been set
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to NULL in the first place because although we have concrete
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evidence that it's happening, we haven't been able to reproduce it.
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Resolves: #783
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#### app_voicemail_odbc: Allow audio to be kept on disk
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Author: George Joseph
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Date: 2024-04-09
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This commit adds a new voicemail.conf option 'odbc_audio_on_disk'
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which when set causes the ODBC variant of app_voicemail to leave
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the message and greeting audio files on disk and only store the
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message metadata in the database. This option came from a concern
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that the database could grow to large and cause remote access
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and/or replication to become slow. In a clustering situation
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with this option, all asterisk instances would share the same
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database for the metadata and either use a shared filesystem
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or other filesystem replication service much more suitable
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for synchronizing files.
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The changes to app_voicemail to implement this feature were actually
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quite small but due to the complexity of the module, the actual
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source code changes were greater. They fall into the following
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categories:
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* Tracing. The module is so complex that it was impossible to
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figure out the path taken for various scenarios without the addition
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of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in
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code that's not related to the functional change. Making this worse
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was the fact that many "if" statements in this module didn't use
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braces. Since the tracing macros add multiple statements, many "if"
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statements had to be converted to use braces.
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* Excessive use of PATH_MAX. Previous maintainers of this module
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used PATH_MAX to allocate character arrays for filesystem paths
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and SQL statements as though they cost nothing. In fact, PATH_MAX
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is defined as 4096 bytes! Some functions had (and still have)
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multiples of these. One function has 7. Given that the vast
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majority of installations use the default spool directory path
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`/var/spool/asterisk/voicemail`, the actual path length is usually
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less than 80 bytes. That's over 4000 bytes wasted. It was the
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same for SQL statement buffers. A 4K buffer for statement that
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only needed 60 bytes. All of these PATH_MAX allocations in the
|
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ODBC related code were changed to dynamically allocated buffers.
|
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The rest will have to be addressed separately.
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* Bug fixes. During the development of this feature, several
|
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pre-existing ODBC related bugs were discovered and fixed. They
|
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had to do with leaving orphaned files on disk, not preserving
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original message ids when moving messages between folders,
|
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not honoring the "formats" config parameter in certain circumstances,
|
||||||
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etc.
|
||||||
|
|
||||||
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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
|
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|
on disk and only store the message metadata in the database.
|
||||||
|
Much more information can be found in the voicemail.conf.sample
|
||||||
|
file.
|
||||||
|
|
||||||
|
|
||||||
|
#### bridge_basic.c: Make sure that ast_bridge_channel is not destroyed while itera..
|
||||||
|
Author: Tinet-mucw
|
||||||
|
Date: 2024-06-13
|
||||||
|
|
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|
Resolves: https://github.com/asterisk/asterisk/issues/768
|
||||||
|
|
||||||
|
#### app_queue: Add option to not log Restricted Caller ID to queue_log
|
||||||
|
Author: Alexei Gradinari
|
||||||
|
Date: 2024-06-12
|
||||||
|
|
||||||
|
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.
|
||||||
|
|
||||||
|
|
||||||
|
#### pbx.c: expand fields width of "core show hints"
|
||||||
|
Author: Alexei Gradinari
|
||||||
|
Date: 2024-06-13
|
||||||
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||||||
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The current width for "extension" is 20 and "device state id" is 20, which is too small.
|
||||||
|
The "extension" field contains "ext"@"context", so 20 characters is not enough.
|
||||||
|
The "device state id" field, for example for Queue pause state contains Queue:"queue_name"_pause_PSJIP/"endpoint", so the 20 characters is not enough.
|
||||||
|
|
||||||
|
Increase the width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters.
|
||||||
|
|
||||||
|
Resolves: #770
|
||||||
|
|
||||||
|
UserNote: The fields width of "core show hints" were increased.
|
||||||
|
The width of "extension" field to 30 characters and
|
||||||
|
the width of the "device state id" field to 60 characters.
|
||||||
|
|
||||||
|
|
||||||
|
#### pjsip: Add PJSIP_PARSE_URI_FROM dialplan function.
|
||||||
|
Author: Sean Bright
|
||||||
|
Date: 2024-06-02
|
||||||
|
|
||||||
|
Various SIP headers permit a URI to be prefaced with a `display-name`
|
||||||
|
production that can include characters (like commas and parentheses)
|
||||||
|
that are problematic for Asterisk's dialplan parser and, specifically
|
||||||
|
in the case of this patch, the PJSIP_PARSE_URI function.
|
||||||
|
|
||||||
|
This patch introduces a new function - `PJSIP_PARSE_URI_FROM` - that
|
||||||
|
behaves identically to `PJSIP_PARSE_URI` except that the first
|
||||||
|
argument is now a variable name and not a literal URI.
|
||||||
|
|
||||||
|
Fixes #756
|
||||||
|
|
||||||
|
|
||||||
|
#### manager.c: Properly terminate `CoreShowChannelMap` event.
|
||||||
|
Author: Sean Bright
|
||||||
|
Date: 2024-06-10
|
||||||
|
|
||||||
|
Fixes #761
|
||||||
|
|
||||||
|
|
||||||
|
#### cli: Show configured cache dir
|
||||||
|
Author: Bastian Triller
|
||||||
|
Date: 2024-06-07
|
||||||
|
|
||||||
|
Since Asterisk 19 it is possible to cache recorded files into another
|
||||||
|
directory [1] [2].
|
||||||
|
Show configured location of cache dir in CLI's core show settings.
|
||||||
|
|
||||||
|
[1] ASTERISK-29143
|
||||||
|
[2] b08427134fd51bb549f198e9f60685f2680c68d7
|
||||||
|
|
||||||
|
|
||||||
|
#### xml.c: Update deprecated libxml2 API usage.
|
||||||
|
Author: Sean Bright
|
||||||
|
Date: 2024-05-23
|
||||||
|
|
||||||
|
Two functions are deprecated as of libxml2 2.12:
|
||||||
|
|
||||||
|
* xmlSubstituteEntitiesDefault
|
||||||
|
* xmlParseMemory
|
||||||
|
|
||||||
|
So we update those with supported API.
|
||||||
|
|
||||||
|
Additionally, `res_calendar_caldav` has been updated to use libxml2's
|
||||||
|
xmlreader API instead of the SAX2 API which has always felt a little
|
||||||
|
hacky (see deleted comment block in `res_calendar_caldav.c`).
|
||||||
|
|
||||||
|
The xmlreader API has been around since libxml2 2.5.0 which was
|
||||||
|
released in 2003.
|
||||||
|
|
||||||
|
Fixes #725
|
||||||
|
|
||||||
|
|
||||||
|
#### cdr_pgsql: Fix crash when the module fails to load multiple times.
|
||||||
|
Author: chrsmj
|
||||||
|
Date: 2024-05-16
|
||||||
|
|
||||||
|
Missing or corrupt cdr_pgsql.conf configuration file can cause the
|
||||||
|
second attempt to load the PostgreSQL CDR module to crash Asterisk via
|
||||||
|
the Command Line Interface because a null CLI command is registered on
|
||||||
|
the first failed attempt to load the module.
|
||||||
|
|
||||||
|
Resolves: #736
|
||||||
|
|
||||||
|
#### asterisk.c: Don't log an error if .asterisk_history does not exist.
|
||||||
|
Author: Sean Bright
|
||||||
|
Date: 2024-05-27
|
||||||
|
|
||||||
|
Fixes #751
|
||||||
|
|
||||||
|
|
||||||
|
#### chan_ooh323: Fix R/0 typo in docs
|
||||||
|
Author: Walter Doekes
|
||||||
|
Date: 2024-05-27
|
||||||
|
|
||||||
|
|
||||||
|
#### bundled_pjproject: Disable UPnP support.
|
||||||
|
Author: Sean Bright
|
||||||
|
Date: 2024-05-24
|
||||||
|
|
||||||
|
Fixes #747
|
||||||
|
|
||||||
|
|
||||||
|
#### file.h: Rename function argument to avoid C++ keyword clash.
|
||||||
|
Author: Sean Bright
|
||||||
|
Date: 2024-05-24
|
||||||
|
|
||||||
|
Fixes #744
|
||||||
|
|
||||||
|
|
||||||
|
#### rtp_engine: add support for multirate RFC2833 digits
|
||||||
|
Author: Mike Bradeen
|
||||||
|
Date: 2024-04-08
|
||||||
|
|
||||||
|
Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs.
|
||||||
|
|
||||||
|
Asterisk currently treats RFC2833 Digits as a single rtp payload type
|
||||||
|
with a fixed bitrate of 8K. This change would expand that to 8, 16,
|
||||||
|
24 and 32K.
|
||||||
|
|
||||||
|
This requires checking the offered rtp types for any of these bitrates
|
||||||
|
and then adding an offer for each (if configured for RFC2833.) DTMF
|
||||||
|
generation must also be changed in order to look at the current outbound
|
||||||
|
codec in order to generate appropriately timed rtp.
|
||||||
|
|
||||||
|
For cases where no outgoing audio has yet been sent prior to digit
|
||||||
|
generation, Asterisk now has a concept of a 'preferred' codec based on
|
||||||
|
offer order.
|
||||||
|
|
||||||
|
On inbound calls Asterisk will mimic the payload types of the RFC2833
|
||||||
|
digits.
|
||||||
|
|
||||||
|
On outbound calls Asterisk will choose the next free payload types starting
|
||||||
|
with 101.
|
||||||
|
|
||||||
|
UserNote: No change in configuration is required in order to enable this
|
||||||
|
feature. Endpoints configured to use RFC2833 will automatically have this
|
||||||
|
enabled. If the endpoint does not support this, it should not include it in
|
||||||
|
the SDP offer/response.
|
||||||
|
|
||||||
|
Resolves: #699
|
||||||
|
|
||||||
Loading…
Reference in new issue