# Release Summary - asterisk-13.0.0-beta2

# Release Summary

### asterisk-13.0.0-beta2

### Date: 2014-09-19

### <asteriskteam@digium.com>

## Table of Contents

1. Summary
2. Contributors
3. Closed Issues
4. Other Changes
5. Diffstat

## Summary

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This release includes new features. For a list of new features that have been included with this release, please see the CHANGES file inside the source package. Since this is new major release, users are encouraged to do extended testing before upgrading to this version in a production environment.

The data in this summary reflects changes that have been made since the previous release, asterisk-13.0.0-beta1.

## Contributors

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This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were closed by commits that went into this release.

### Coders

### Testers

### Reporters

21 rmudgett
20 mmichelson
16 mjordan
13 jrose
10 file
10 gtjoseph
7 kmoore
4 jcolp
3 wdoekes
2 Jeremy Laine
2 seanbright
2 sgriepentrog
2 sruffell
1 cloos
1 dlee
1 Elazar Broad
1 elguero
1 newtonr
1 wedhorn

2 George Joseph
1 Damien Wedhorn
1 David Herselman
1 Deepak Singh Rawat
1 dimitripietro
1 elguero
1 Kilburn
1 Samuel Galarneau
1 sruffell
1 Tony Lewis
1 wdoekes

7 mjordan
3 sruffell
2 mmichelson
2 sharky
1 amohod
1 ateks
1 bbs2web
1 dafi
1 dimitripietro
1 dsr
1 Each
1 ebroad
1 edvinv
1 falves11
1 jideliov
1 jrose
1 krandonbruse
1 maddog
1 pnlarsson
1 proftech
1 rmudgett
1 RomanSk
1 sgalarneau
1 sgriepentrog
1 slavon
1 wdoekes
1 xrobau

## Closed Issues

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This is a list of all issues from the issue tracker that were closed by changes that went into this release.

### Category: . I did not set the category correctly.

ASTERISK-24147: ARI: channel hangup crashes asterisk process
Revision: 421880
Reporter: edvinv
Coders: jrose

### Category: Applications/app_controlplayback

ASTERISK-24229: ARI: playback of sounds implicitly answers channel, preventing early media playback
Revision: 421696
Reporter: mjordan
Coders: mjordan

### Category: Applications/app_dial

ASTERISK-24225: Dial option z is broken
Revision: 421235
Reporter: dimitripietro
Testers: dimitripietro
Coders: rmudgett

### Category: Applications/app_meetme

ASTERISK-24234: app_meetme: Crash on conference shutdown due to NULL channel passed to meetme_stasis_generate_msg()
Revision: 421273
Reporter: sruffell
Testers: sruffell
Coders: mjordan

### Category: Applications/app_mixmonitor

ASTERISK-24027: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up
Revision: 420940
Reporter: mjordan
Coders: jrose
ASTERISK-24027: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up
Revision: 421187
Reporter: mjordan
Coders: jrose

### Category: CDR/General

ASTERISK-24237: CDR: FRACK With PJSIP blonde transfer.
Revision: 423530
Reporter: rmudgett
Coders: jrose
ASTERISK-24241: crash: CDRs recursively attempt to update Party B information in a multi-party bridge, overrunning the stack
Revision: 422716
Reporter: dsr
Testers: Deepak Singh Rawat
Coders: mjordan
ASTERISK-24254: CDRs: Application/args/dialplan CEP updated during dial operation
Revision: 422719
Reporter: mjordan
Testers: Tony Lewis
Coders: mjordan

### Category: Channels/chan_iax2

ASTERISK-23767: [patch] Dynamic IAX2 registration stops trying if ever not able to resolve
Revision: 422276
Reporter: bbs2web
Testers: David Herselman, elguero
Coders: elguero
ASTERISK-24265: segfault in asterisk when try to make call to IAX
Revision: 423524
Reporter: dafi
Coders: jrose

### Category: Channels/chan_pjsip

ASTERISK-24143: pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK
Revision: 421956
Reporter: Each
Coders: jcolp
ASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP engine
Revision: 422542
Reporter: mjordan
Coders: mmichelson

### Category: Channels/chan_sip/General

ASTERISK-24178: [patch]fromdomainport used even if not set
Revision: 421720
Reporter: ebroad
Coders: Elazar Broad

### Category: Channels/chan_sip/Messaging

ASTERISK-24301: Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk
Revision: 423372
Reporter: mjordan
Coders: mmichelson

### Category: Channels/chan_sip/WebSocket

ASTERISK-23997: chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer
Revision: 421911
Reporter: slavon
Coders: jcolp

### Category: Core/Configuration

ASTERISK-24231: crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable
Revision: 422985
Reporter: pnlarsson
Coders: jrose

### Category: Core/ManagerInterface

ASTERISK-24331: Unexpected Errors in Asterisk Manager Interface Output
Revision: 423284
Reporter: xrobau
Testers: George Joseph
Coders: gtjoseph

### Category: Core/PBX

ASTERISK-24249: SIP debugs do not stop
Revision: 423069
Reporter: amohod
Coders: wdoekes

### Category: Documentation

ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
Revision: 422374
Reporter: sharky
Coders: Jeremy Laine
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
Revision: 422379
Reporter: sharky
Coders: Jeremy Laine

### Category: General

ASTERISK-24032: Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined
Revision: 421230
Reporter: maddog
Testers: Kilburn, wdoekes
Coders: cloos

### Category: Resources/res_agi

ASTERISK-24027: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up
Revision: 420940
Reporter: mjordan
Coders: jrose
ASTERISK-24027: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up
Revision: 421187
Reporter: mjordan
Coders: jrose

### Category: Resources/res_ari

ASTERISK-24043: ARI /continue fails to actually continue into the dialplan
Revision: 421423
Reporter: krandonbruse
Coders: jrose
ASTERISK-24229: ARI: playback of sounds implicitly answers channel, preventing early media playback
Revision: 421696
Reporter: mjordan
Coders: mjordan
ASTERISK-24264: ARI: Adding a channel to a holding bridge automatically starts MOH
Revision: 422504
Reporter: sgalarneau
Testers: Samuel Galarneau
Coders: mjordan

### Category: Resources/res_ari_bridges

ASTERISK-24264: ARI: Adding a channel to a holding bridge automatically starts MOH
Revision: 422504
Reporter: sgalarneau
Testers: Samuel Galarneau
Coders: mjordan

### Category: Resources/res_ari_playbacks

ASTERISK-24229: ARI: playback of sounds implicitly answers channel, preventing early media playback
Revision: 421696
Reporter: mjordan
Coders: mjordan

### Category: Resources/res_fax

ASTERISK-24301: Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk
Revision: 423372
Reporter: mjordan
Coders: mmichelson

### Category: Resources/res_hep_rtcp

ASTERISK-24236: res_hep_rtcp: Module incorrectly depends on pjsip
Revision: 421065
Reporter: mjordan
Testers: Damien Wedhorn
Coders: mjordan

### Category: Resources/res_musiconhold

ASTERISK-22252: res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks
Revision: 421788
Reporter: wdoekes
Coders: jrose
ASTERISK-24019: When a Music On Hold stream starts it restarts at beginning of file.
Revision: 421979
Reporter: ateks
Coders: rmudgett

### Category: Resources/res_pjsip

ASTERISK-24161: PJSIPShowEndpoint gives inaccurate count of list items
Revision: 423284
Reporter: mmichelson
Testers: George Joseph
Coders: gtjoseph

### Category: Resources/res_pjsip_endpoint_identifier_ip

ASTERISK-24290: Endpoint identifier match value fails to parse when CIDR network format is specified
Revision: 423425
Reporter: proftech
Coders: jrose

### Category: Resources/res_pjsip_nat

ASTERISK-23634: With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls
Revision: 423152
Reporter: RomanSk
Coders: jcolp

### Category: Resources/res_pjsip_pubsub

ASTERISK-24136: Security: Crash in Asterisk's PJSIP code when subscribing to an event with an unexpected body type
Revision: 423348
Reporter: mmichelson
Coders: mmichelson
ASTERISK-24181: RLS: Large lists don't get sent because they exceed the PJSIP message length limit
Revision: 422851
Reporter: jrose
Coders: mmichelson

### Category: Resources/res_pjsip_sdp_rtp

ASTERISK-23994: res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname
Revision: 421797
Reporter: falves11
Coders: mmichelson

### Category: Resources/res_pjsip_transport_websocket

ASTERISK-24143: pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK
Revision: 421956
Reporter: Each
Coders: jcolp

### Category: Resources/res_rtp_asterisk

ASTERISK-23577: res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL
Revision: 423152
Reporter: jideliov
Coders: jcolp
ASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP engine
Revision: 422542
Reporter: mjordan
Coders: mmichelson

### Category: Tests/testsuite

ASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP engine
Revision: 422542
Reporter: mjordan
Coders: mmichelson

### Category: Utilities/aelparse

ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
Revision: 422374
Reporter: sharky
Coders: Jeremy Laine
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
Revision: 422379
Reporter: sharky
Coders: Jeremy Laine

## Commits Not Associated with an Issue

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This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.

Revision | Author | Summary | Issues Referenced
420837 | rmudgett | res/stasis/command.c: Fix recent commit using spaces instead of tabs. |
420856 | file | app_voicemail: Fix the "test_voicemail_vm_info" unit test. |
420879 | rmudgett | res_stasis_snoop.c: Fix off nominial exit path leaving Snoop channel locked and not hungup. |
420881 | rmudgett | chan_sip: Fix type mismatch when the format is changed. |
420899 | wdoekes | logger: Don't store verbose-magic in the log files. |
420919 | kmoore | AMI: Improve documentation for Status action |
420950 | kmoore | PJSIP: Prevent crash no-URI contacts |
420957 | rmudgett | res_pjsip_send_to_voicemail.c: Fix svn file properties. |
420992 | rmudgett | channel_internal_api.c: Replace some code with ao2_replace(). |
421010 | rmudgett | ARI: Originate to app local channel subscription code optimization. |
421042 | mjordan | cel: Make sure channels in extra fields include their unique IDs as well |
421062 | mjordan | main/file: Move test event to emit PLAYBACK event more consistently |
421166 | mjordan | app_voicemail/app: Remove test events that were duplicated by r421059 |
421210 | file | res_http_websocket: Include query parameters in client connection requests. |
421311 | mjordan | res/ari/resource_channels: Don't return allocation failure on failed function |
421312 | mjordan | res/ari/resource_channels: Fix compilation issue |
421337 | gtjoseph | func_config: Change 'Not Found' message from ERROR to DEBUG |
421376 | wedhorn | Skinny: Fixup compile warning for non dev-mode. |
421403 | rmudgett | chan_pjsip: Fix attended transfer connected line name update. |
421445 | kmoore | AMI Docs: Fix Status channel parameter optionality |
421448 | mmichelson | Fix compilation error on certain versions of GCC. |
421488 | mmichelson | Alter documentation for callerid_privacy to use correct values. |
421538 | kmoore | Stasis: Add information to blind transfer event |
421566 | mmichelson | Move evaluation of set_var options in pjsip to the end of channel initialization. |
421585 | mmichelson | Set the role for inbound subscriptions correctly. |
421616 | rmudgett | cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled. |
421645 | rmudgett | chan_pjsip: Update media translation paths when new SDP negotiated. |
421783 | mmichelson | Improve consistency of party ID privacy usage. |
421790 | mmichelson | Let's try checking the name and number, instead of the name twice. |
421794 | mmichelson | Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge. |
421802 | rmudgett | res_musiconhold.c: Remove obsolete REF_DEBUG code. |
421860 | mjordan | main/message: Add a new-line to a DEBUG message |
421882 | mmichelson | Fix a locking inversion in MixMonitor. |
421932 | file | res_pjsip_transport_websocket: Ensure secure Websocket clients can be called. |
421945 | file | res_pjsip_transport_websocket: Fix a progressive memory growth. |
422037 | rmudgett | res_musiconhold.c: Release any format refs before memset(). |
422071 | mmichelson | Fix race condition in the scheduler when deleting a running entry. |
422091 | gtjoseph | confbridge: Make kick, mute and unmute handle channel targets consistently. |
422154 | kmoore | CallerID: Fix parsing of malformed callerid |
422177 | gtjoseph | confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events |
422200 | rmudgett | sched: Fix typo and whitespace change. |
422215 | rmudgett | res/res_pjsip/pjsip_options.c: Eliminate excessive RAII_VAR usage. |
422239 | mmichelson | Fix bug that did not allow for multiple batched RLS notifications to be sent. |
422256 | rmudgett | Added ConfBridge AMI event note to UPGRADE.txt. |
422296 | mjordan | LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP |
422359 | sgriepentrog | The assertion that peer was not found on final event |
422442 | gtjoseph | manager: Make WaitEvent action respect eventfilters |
422445 | gtjoseph | confbridge: Add Duration to ConfbridgeList event |
422507 | mjordan | main/cli: Do not attempt to show CDR data for internal channels |
422558 | file | res_pjsip_transport_websocket: Fix crash when the Contact header is not a URI. |
422631 | jrose | Manager: Require read permission for SYSTEM in order to send FullyBooted |
422646 | kmoore | Menuselect: Fix incorrect enabling on failed deps |
422661 | rmudgett | devicestate.c: Minor tweaks |
422665 | jrose | Call IDs: Fix appearance of call ID in core show channels when NULL |
422684 | jrose | Dial API: Add a dial option to indicate the dialed channel will replace dialer |
422700 | rmudgett | func_channel.c: Add missing locking to some CHANNEL() requests. |
422747 | file | res_pjsip_sdp_rtp: Fix retrieval of "ice-pwd" attribute if in session and not media stream. |
422767 | mjordan | main/rtp_engine: Format NTP timestamps as unsigned ints |
422770 | mjordan | main/cdr: Copy over location information during a fork |
422836 | jrose | res_pjsip_pubsub: Check supported headers for eventlist when subscribing to |
422853 | mmichelson | Add sample configuration for resource lists. |
422855 | mmichelson | Add note about configuring list_items on a single line. |
422883 | newtonr | Sounds/BuildSystem: Modifications to include new releases and Japanese language. |
422901 | seanbright | pjsip/config_auth.c: Add missing whitespace to log messages. |
422905 | gtjoseph | config: bug: fix truncation of included config files on permissions error |
422965 | mmichelson | Remove undocumented default behavior of ast_play_and_record_full acceptdtmf. |
423129 | wdoekes | contrib: Fix verifyi typo in alembic DB script ps_transport table. |
423173 | file | res_pjsip_session: Fix usage of wrong memory pool when creating local SDP. |
423209 | file | res_rtp_asterisk: Fix building when pjproject is not used. |
423212 | file | res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning. |
423255 | file | res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered. |
423279 | gtjoseph | config: bug: Fix SEGV in ast_category_insert when matching category isn't found |
423281 | dlee | Only install dahdi_span_config_hook if DAHDI is enabled |
423414 | mmichelson | Add API call to determine if format capability structure is "empty". |
423418 | rmudgett | astobj2.c/refcounter.py: Fix to deal with invalid object refs. |
423423 | rmudgett | bridge_softmix.c: Made use ao2_replace() instead of the inline equivalent. |
423462 | mmichelson | Add subscription state test events. |
423478 | gtjoseph | utils: Create ast_strsep function that ignores separators inside quotes |
423482 | seanbright | res_pjsip: Don't require a password when doing userpass authentication. |
423504 | kmoore | PJSIP: Prevent T38 framehook being put on wrong channel |
423561 | rmudgett | res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated. |

## Diffstat Results

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This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.

    LICENSE                                                                      |    2
    Makefile                                                                     |   10
    UPGRADE.txt                                                                  |   19
    apps/app_chanspy.c                                                           |    2
    apps/app_confbridge.c                                                        |  265 ++-
    apps/app_dial.c                                                              |    2
    apps/app_macro.c                                                             |    7
    apps/app_meetme.c                                                            |    8
    apps/app_mixmonitor.c                                                        |    7
    apps/app_stack.c                                                             |   35
    apps/app_voicemail.c                                                         |   21
    apps/confbridge/confbridge_manager.c                                         |   81 +
    bridges/bridge_softmix.c                                                     |   13
    channels/chan_iax2.c                                                         |   45
    channels/chan_pjsip.c                                                        |  102 -
    channels/chan_sip.c                                                          |   33
    channels/chan_skinny.c                                                       |   10
    configs/samples/pjsip.conf.sample                                            |   50
    configs/samples/sip.conf.sample                                              |    4
    configure.ac                                                                 |    4
    contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py |   29
    contrib/scripts/refcounter.py                                                |   80 -
    doc/aelparse.8                                                               |   28
    doc/smsq.8                                                                   |  146 ++
    funcs/func_channel.c                                                         |   28
    funcs/func_config.c                                                          |    2
    include/asterisk/channel.h                                                   |   43
    include/asterisk/config.h                                                    |    5
    include/asterisk/dial.h                                                      |    1
    include/asterisk/file.h                                                      |    7
    include/asterisk/format_cap.h                                                |    9
    include/asterisk/framehook.h                                                 |    6
    include/asterisk/res_pjsip.h                                                 |    2
    include/asterisk/res_pjsip_pubsub.h                                          |   23
    include/asterisk/res_pjsip_session.h                                         |   35
    include/asterisk/rtp_engine.h                                                |    4
    include/asterisk/sched.h                                                     |    2
    include/asterisk/stasis_app_impl.h                                           |   25
    include/asterisk/stasis_bridges.h                                            |   10
    include/asterisk/strings.h                                                   |   60
    include/asterisk/uri.h                                                       |    2
    include/asterisk/utils.h                                                     |    9
    main/app.c                                                                   |    7
    main/astobj2.c                                                               |   10
    main/bridge.c                                                                |   63
    main/bridge_after.c                                                          |    4
    main/bridge_channel.c                                                        |   15
    main/callerid.c                                                              |   63
    main/cdr.c                                                                   |   22
    main/cel.c                                                                   |   27
    main/channel.c                                                               |  159 +-
    main/channel_internal_api.c                                                  |   52
    main/cli.c                                                                   |    6
    main/config.c                                                                |  132 +
    main/devicestate.c                                                           |    5
    main/dial.c                                                                  |    7
    main/dns.c                                                                   |    3
    main/file.c                                                                  |    9
    main/format_cache.c                                                          |    3
    main/format_cap.c                                                            |   15
    main/framehook.c                                                             |   19
    main/logger.c                                                                |   24
    main/manager.c                                                               |   35
    main/message.c                                                               |    2
    main/pbx.c                                                                   |    5
    main/rtp_engine.c                                                            |    4
    main/sched.c                                                                 |   49
    main/stasis_bridges.c                                                        |   28
    main/stasis_channels.c                                                       |  219 +++
    main/translate.c                                                             |    5
    main/uri.c                                                                   |    2
    main/utils.c                                                                 |   83 +
    makeopts.in                                                                  |    1
    menuselect/menuselect.c                                                      |    2
    res/ari/ari_model_validators.c                                               |    9
    res/ari/ari_model_validators.h                                               |    1
    res/ari/resource_channels.c                                                  |   17
    res/res_fax_spandsp.c                                                        |   19
    res/res_hep_rtcp.c                                                           |    2
    res/res_http_websocket.c                                                     |   26
    res/res_musiconhold.c                                                        |   33
    res/res_pjsip.c                                                              |   10
    res/res_pjsip/config_auth.c                                                  |   14
    res/res_pjsip/config_transport.c                                             |   18
    res/res_pjsip/location.c                                                     |    2
    res/res_pjsip/pjsip_configuration.c                                          |    6
    res/res_pjsip/pjsip_options.c                                                |  173 +-
    res/res_pjsip_caller_id.c                                                    |   94 -
    res/res_pjsip_dialog_info_body_generator.c                                   |    1
    res/res_pjsip_diversion.c                                                    |    1
    res/res_pjsip_endpoint_identifier_ip.c                                       |   62
    res/res_pjsip_exten_state.c                                                  |    8
    res/res_pjsip_mwi.c                                                          |   13
    res/res_pjsip_mwi_body_generator.c                                           |    1
    res/res_pjsip_notify.c                                                       |    8
    res/res_pjsip_pidf_body_generator.c                                          |    1
    res/res_pjsip_pubsub.c                                                       |  166 ++
    res/res_pjsip_sdp_rtp.c                                                      |   31
    res/res_pjsip_session.c                                                      |   72 -
    res/res_pjsip_t38.c                                                          |   13
    res/res_pjsip_transport_websocket.c                                          |   46
    res/res_pjsip_xpidf_body_generator.c                                         |    2
    res/res_rtp_asterisk.c                                                       |  704 ++++++----
    res/res_stasis.c                                                             |   30
    res/res_stasis_answer.c                                                      |    2
    res/res_stasis_playback.c                                                    |   20
    res/res_stasis_recording.c                                                   |   20
    res/res_stasis_snoop.c                                                       |    4
    res/stasis/app.c                                                             |   31
    res/stasis/command.c                                                         |   41
    res/stasis/command.h                                                         |    9
    res/stasis/control.c                                                         |   81 -
    res/stasis/messaging.h                                                       |    2
    res/stasis/stasis_bridge.c                                                   |   28
    rest-api/api-docs/events.json                                                |    5
    sounds/Makefile                                                              |    7
    sounds/sounds.xml                                                            |   27
    tests/test_callerid.c                                                        |  165 ++
    tests/test_cel.c                                                             |   21
    tests/test_strings.c                                                         |   80 +
    tests/test_utils.c                                                           |   98 +
    121 files changed, 3531 insertions(+), 1049 deletions(-)