The release of Asterisk 15.1.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Improvements made in this release:
-----------------------------------
| [patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE | (Reported by David J. Pryke) | |
| alembic: Add support for Microsoft SQL server | (Reported by Florian Floimair) | |
| [patch] libsrtp-2.1.x support | (Reported by Alexander Traud) | |
| Enable CHANNEL function to get from and to tag from SIP Headers | (Reported by Andre Nazario) | |
| Google OAuth 2.0 support for XMPP / Motif | (Reported by Andrey) | |
| Support for GMIME 3.0 | (Reported by Tzafrir Cohen) | |
| [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip | (Reported by Torrey Searle) |
Bugs fixed in this release:
-----------------------------------
| res_xmpp: Crash if OAuth 2.0 is used before curl is loaded | (Reported by Ronald Raikes) | |
| ARI: Node ARI client broken in latest versions of 13 and 14 | (Reported by Benjamin Keith Ford) | |
| res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be. | (Reported by dtryba) | |
| res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs | (Reported by dtryba) | |
| cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured | (Reported by Tzafrir Cohen) | |
| Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate | (Reported by Allen Ford) | |
| res_pjsip: Loss of SIP registrations causing unavailable endpoints | (Reported by Richard Mudgett) | |
| res_ari: Memory leaks in ARI when using Content-Type: application/json | (Reported by David Hajek) | |
| chan_sip: tcpbind uses wrong source address | (Reported by Ksenia) | |
| [patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS | (Reported by Alexander Traud) | |
| vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. | (Reported by Corey Farrell) | |
| [patch] app_queue: Music On Hold for real-time queues is not reset to default | (Reported by Nathan Bruning) | |
| res_pjsip_mwi: uninitialized value from ast_strings_match | (Reported by Corey Farrell) | |
| Status of RFC 3323 and PJSIP | (Reported by dtryba) | |
| [patch] False positive busy checks when icalendar's recurrence-id mechanism is involved | (Reported by Benoît Dereck-Tricot) | |
| app_queue: does its check-makeannouncement-logic twice each head-caller-loop | (Reported by Stefan Engström) | |
| Problem with expires on pjsip / outbound-publish | (Reported by Cyrille Demaret) | |
| Contact is improperly translated after d178f497 | (Reported by Sean Bright) | |
| Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes) | (Reported by Ross Beer) | |
| chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller | (Reported by lvl) | |
| A codeblock that maintains a bug,but maybe the codeblock will never run | (Reported by Huangyx) | |
| bridge: Renegotiate if source stream changes. | (Reported by Joshua Colp) | |
| Realtime config fail with PostgreSQL version before 9.1 | (Reported by Rodrigo Ramirez Norambuena) | |
| res_pjsip_session: Crashes after sending PRACK and receiving 200 OK | (Reported by Daniel Heckl) | |
| [pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36 | (Reported by Daniel Heckl) | |
| bridge_native_rtp: half-way direct media when using early bridging | (Reported by Jean Aunis - Prescom) | |
| SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0 | (Reported by Marcello Ceschia) | |
| RTCP needs better packet validation to resist port scans. | (Reported by Richard Mudgett) | |
| RTP: One way audio with direct media and strictrtp=yes. | (Reported by Richard Mudgett) | |
| Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability | (Reported by Ross Beer) | |
| module reload res_calendar.so does not reload everything in calendar.conf | (Reported by Jesper) | |
| res_calendar does not process CalDAV from Owncloud [fix included] | (Reported by Stefan Gofferje) | |
| res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured. | (Reported by Jesper) | |
| RTP Multicast of L16 (type 10): Asterisk and wireshark disagree | (Reported by Tzafrir Cohen) | |
| [patch]external_media_address and external_signaling_address don't always honor localnet | (Reported by Walter Doekes) | |
| res_smdi: convert to astobj2 | (Reported by Corey Farrell) | |
| chan_sip: Asterisk crashing when subscription doesn't get set | (Reported by Bryan Walters) | |
| SDP origin attribute modified when issuing re-INVITE because of directmedia=yes | (Reported by saghul) | |
| CDR: CDR(start,u) function won't work in cdr_custom config | (Reported by Jacek Konieczny) | |
| alembic: prune_on_boot fix erroneous | (Reported by Florian Floimair) | |
| When in queue on g722 with interruptions, music on hold can get stuck and no longer play | (Reported by Jens T.) | |
| nat/external_media settings ignored in 14.4.1 | (Reported by Christopher van de Sande) | |
| PJSIP external_media_address ignored if no local_net options are provided | (Reported by Matt Jordan) | |
| Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive | (Reported by Ross Beer) | |
| Crash when freeing dtls_cfg->cafile | (Reported by Richard Kenner) | |
| ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c | (Reported by Tzafrir Cohen) | |
| libc segfault upon entry into app_directory | (Reported by David Moore) | |
| Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash | (Reported by Ross Beer) | |
| core: ast_safe_system command injection possible. | (Reported by Corey Farrell) | |
| res_rtp_asterisk: Media can be hijacked even with strict RTP enabled | (Reported by Joshua Colp) | |
| res_rtp_asterisk: Allow remote SSRC to change due to renegotiation | (Reported by Joshua Colp) | |
| Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name | (Reported by James Terhune) | |
| core: Don't queue up multiple video update frames. | (Reported by Joshua Colp) | |
| app_minivm fails to clean up mkstemp files | (Reported by Walter Doekes) | |
| several filename bugs in Record() application | (Reported by klaus3000) | |
| alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table | (Reported by Florian Floimair) | |
| Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used | (Reported by Torrey Searle) | |
| When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used. | (Reported by Jim Van Meggelen) | |
| When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored | (Reported by Eelco Brolman) | |
| bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU | (Reported by Richard Mudgett) | |
| [patch] app_queue: Wrong queue stat calculation | (Reported by sungtae kim) | |
| XMPP OAuth not working due to inverted logic | (Reported by Michael Kuron) | |
| res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical | (Reported by Mark Thompson) | |
| If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed | (Reported by Seán C. McCord) | |
| manager: hook event is not being raised | (Reported by Kevin Harwell) | |
| Either asterisk or pjproject isn't re-using tcp connections (again) | (Reported by George Joseph) | |
| IPv6 receive address in message doesn't include brackets | (Reported by Scott Griepentrog) | |
| [patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used | (Reported by Torrey Searle) | |
| Asymmetric codecs when asymmetric_rtp_codec=no | (Reported by Jesse Ross) | |
| Make --with-pjproject-bundled the default for Asterisk 15 | (Reported by George Joseph) | |
| RTP session is not fully destroyed on channel hangup | (Reported by Matt Jordan) | |
| bridge: Crash when mapping streams | (Reported by Joshua Colp) | |
| channel: requester leaks joint_cap on success. | (Reported by Corey Farrell) | |
| res_pjsip_session: Handling of 'msid' is incorrect | (Reported by Kevin Harwell) | |
| res_pjsip: parse/add msid attribute when webrtc is enabled | (Reported by Kevin Harwell) | |
| Asterisk 15.0.0-Beta1 does not compile | (Reported by Ira Emus) | |
| res_pjsip: PJSIP presence - missing braces around the status element in XML | (Reported by Abraham Liebsch) | |
| Asterisk won't compile on Fedora 26 with devmode enabled. | (Reported by Corey Farrell) |
New Features made in this release:
-----------------------------------
| [patch]AMI : Add CancelAtxfer Action | (Reported by Thomas Sevestre) |
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
Thank you for your continued support of Asterisk!
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