Re: [asterisk-dev] Call unhold/topology change indication order

2022-05-12 Thread Joshua C. Colp 
On Thu, May 12, 2022 at 12:06 PM Fridrich Maximilian 
wrote:

> Hi,
>
> I think I have found out why the indication order on hold/unhold matters:
>
> AST_CONTROL_HOLD/UNHOLD only cares about the audio stream and does not
> touch
> the topology of any other streams. So when Asterisk receives an SDP with
> audio
> and video and both are sendonly, chan_pjsip indicates AST_CONTROL_HOLD for
> the audio stream and AST_CONTROL_STREAM_TOPOLOGY_REQUEST_CHANGE for the
> video
> stream.
>
> If AST_CONTROL_HOLD is indicated after
> AST_CONTROL_STREAM_TOPOLOGY_REQUEST_CHANGE, it still has "outdated"
> topology
> information as it only cares about the default audio stream. So after the
> stream topology is changed, it is "overwritten" by the outdated topology
> from
> the hold/unhold indication.
>
> To resolve the issue, the topology request change needs to check if this is
> also a hold/unhold. If it is, there are two option:
>
> 1. Ensure that it executes after the hold/unhold indication
> 2. Ensure that the hold/unhold indication receives the updated topology
>
> I'm stuck on implementing either of those solutions. I think the place we
> need
> to work on is in bridge_channel.c:bridge_handle_trip() before the call to
> stream_topology_changed(). In bridge_handle_trip() we might still have a
> chance
> to interact with the other channel(s) in the bridge.
>
> Does anyone have any idea on how to proceed?
>

Not off the top of my head. I will try to give this some thought but I
don't know if/when I'll really have any thought, it's not something that
can really be answered without digging in deeply and refreshing memory on
the entire view of everything.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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[asterisk-dev] Asterisk 19.4.0 Now Available

2022-05-12 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
19.4.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.4.0 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:

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)

New Features made in this release:
---
 * ASTERISK-29931 - Option to allow a user to not hear the join
  sound on enter but everyone else can
  (Reported by Michael
  Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
  cardinality of keys at prefix
  (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
  without device state
  (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function

  (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash
  events
  (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
  camera available
  (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early
  media
  (Reported by Michael Auracher)
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
  when Picking Up Dahdi Call On Hold
  (Reported by Josh
  Alberts)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
  encryption with missing secrets
  (Reported by N A)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
  idempotent on dahdi restart
  (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
  are enabled are always recompiled
  (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
  variables when channel is NULL
  (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
  with "r" or "R" flags. (documentation bug)
  (Reported by
  Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
  for "disable console colorization"
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
  small for max number of groups
  (Reported by N A)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
  not prevented
  (Reported by N A)
 * ASTERISK-29843 - Session timers get removed on UPDATE
 
  (Reported by Mark Petersen)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
  even if early_media already enabled
  (Reported by Mark
  Petersen)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
  UPDATE
  (Reported by Mark Petersen)
 * ASTERISK-29253 - Incorrect bridging on transfer
 
  (Reported by Yury Kirsanov)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data

  (Reported by N A)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
  async_operations is greater than 1
  (Reported by Ross Beer)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
  variable with new keyword
  (Reported by Jasper
  Hafkenscheid)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
  of SDP attributes
  (Reported by Josh Hogan)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
  database columns
  (Reported by Gregory Massel)
 * ASTERISK-26689 - res_pjsip_sdp_rtp: 183 Session in Progress.
  Disconnecting channel for lack of RTP activity
  (Reported
  by Dmitriy Serov)
 * ASTERISK-29929 - res_pjsip_sdp_rtp: Disconnecting channel for
  lack of RTP activity in one way sessions
  (Reported by
  Boris P. Korzun)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
 
  (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
   
  (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
  context (AST_PBX_MAX_STACK - 1)
  (Reported by Tzafrir
  Cohen)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn't available
  (Reported by Stefan Ruijsenaars)
 * ASTERISK-29988 - REGRESSION: The build process is requiring

[asterisk-dev] Asterisk 18.12.0 Now Available

2022-05-12 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
18.12.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.12.0 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:

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)

New Features made in this release:
---
 * ASTERISK-29931 - Option to allow a user to not hear the join
  sound on enter but everyone else can
  (Reported by Michael
  Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
  cardinality of keys at prefix
  (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
  without device state
  (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash
  events
  (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function

  (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
  camera available
  (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early
  media
  (Reported by Michael Auracher)
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
  when Picking Up Dahdi Call On Hold
  (Reported by Josh
  Alberts)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
  idempotent on dahdi restart
  (Reported by N A)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
  encryption with missing secrets
  (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
  are enabled are always recompiled
  (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
  variables when channel is NULL
  (Reported by N A)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
  small for max number of groups
  (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
  with "r" or "R" flags. (documentation bug)
  (Reported by
  Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
  for "disable console colorization"
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-29843 - Session timers get removed on UPDATE
 
  (Reported by Mark Petersen)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
  not prevented
  (Reported by N A)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
  UPDATE
  (Reported by Mark Petersen)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
  even if early_media already enabled
  (Reported by Mark
  Petersen)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data

  (Reported by N A)
 * ASTERISK-29253 - Incorrect bridging on transfer
 
  (Reported by Yury Kirsanov)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
  async_operations is greater than 1
  (Reported by Ross Beer)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
  variable with new keyword
  (Reported by Jasper
  Hafkenscheid)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
  database columns
  (Reported by Gregory Massel)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
  of SDP attributes
  (Reported by Josh Hogan)
 * ASTERISK-26689 - res_pjsip_sdp_rtp: 183 Session in Progress.
  Disconnecting channel for lack of RTP activity
  (Reported
  by Dmitriy Serov)
 * ASTERISK-29929 - res_pjsip_sdp_rtp: Disconnecting channel for
  lack of RTP activity in one way sessions
  (Reported by
  Boris P. Korzun)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
 
  (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
   
  (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
  context (AST_PBX_MAX_STACK - 1)
  (Reported by Tzafrir
  Cohen)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn't available
  (Reported by Stefan Ruijsenaars)
 * ASTERISK-29988 - REGRESSION: The build process is requiring

[asterisk-dev] Asterisk 16.26.0 Now Available

2022-05-12 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
16.26.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.26.0 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:

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)

New Features made in this release:
---
 * ASTERISK-29931 - Option to allow a user to not hear the join
  sound on enter but everyone else can
  (Reported by Michael
  Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
  cardinality of keys at prefix
  (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
  without device state
  (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash
  events
  (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function

  (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
  when Picking Up Dahdi Call On Hold
  (Reported by Josh
  Alberts)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
  idempotent on dahdi restart
  (Reported by N A)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
  encryption with missing secrets
  (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
  are enabled are always recompiled
  (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
  variables when channel is NULL
  (Reported by N A)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
  small for max number of groups
  (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
  with "r" or "R" flags. (documentation bug)
  (Reported by
  Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
  for "disable console colorization"
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-29843 - Session timers get removed on UPDATE
 
  (Reported by Mark Petersen)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
  not prevented
  (Reported by N A)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
  UPDATE
  (Reported by Mark Petersen)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
  even if early_media already enabled
  (Reported by Mark
  Petersen)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data

  (Reported by N A)
 * ASTERISK-29253 - Incorrect bridging on transfer
 
  (Reported by Yury Kirsanov)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
  async_operations is greater than 1
  (Reported by Ross Beer)
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
  camera available
  (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early
  media
  (Reported by Michael Auracher)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
  of SDP attributes
  (Reported by Josh Hogan)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
  variable with new keyword
  (Reported by Jasper
  Hafkenscheid)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
  database columns
  (Reported by Gregory Massel)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
 
  (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
   
  (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
  context (AST_PBX_MAX_STACK - 1)
  (Reported by Tzafrir
  Cohen)
 * ASTERISK-29988 - REGRESSION: The build process is requiring
  xmllint or xmlstarlet ro be installed when it shouldn't
 
  (Reported by George Joseph)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn't available
  (Reported by Stefan Ruijsenaars)
 * ASTERISK-29895 - chan_iax2: Fix misaligned spacing in iax2
  show netstats printout
  (Reported by N A)
 * ASTERISK-29939 - agi: Fix xmldoc bug with set music
 
  (Reported by N A)
 *