Re: [asterisk-users] Opus codec in codecs.conf

2016-10-25 Thread Igor Goncharovsky 
Hello,

George, thank you for pointing this, but there is other question. It is not
clear for some parameters names, what is possible values?

For example this parameters:
packet_loss
max_bandwidth
signal
application

Is there example of configured opus with full set of parameters?


2016-10-25 18:42 GMT+06:00 George Joseph :

>
>
> On Mon, Oct 24, 2016 at 6:54 PM, Igor Goncharovsky <
> igor.goncharov...@gmail.com> wrote:
>
>> Hello,
>>
>> I am trying to configure new opus codec in asterisk 14, but unable to
>> find any examples of codecs.conf settings for this codec.
>>
>> All I am trying to do - setup peer with using opus in narrow band mode
>> (8kHz sampling rate). Does anybody know how to configure chan_opus?
>>
>>
> If you run "config show help condec_opus opus" from teh Asterisk command
> line, you'll get a list of the configuration options
>
> pbx1*CLI> config show help codec_opus opus
> opus: [category !~ /.?/]
>
> Codec opus module for Asterisk options
>
> type  -- Must be of type 'opus'
> sample_rate   -- Codec's sample rate.
> packet_loss   -- Encoder's packet loss percentage.
>
> complexity-- Encoder's computational complexity.
>
> max_bandwidth -- Encoder's maximum bandwidth allowed.
>
> signal-- Encoder's signal type.
> application   -- Encoder's application type.
> max_playback_rate -- Encoder's maximum playback rate.
>
> max_ptime -- Encoder's maximum packetization rate.
>
> ptime -- Encoder's packetization rate.
> bitrate   -- Encoder's bit rate.
> cbr   -- Encoder's constant bit rate value.
> fec   -- Encoder's forward error correction value.
> dtx   -- Encoder's discontinuous transmission value.
>
>
>
>> --
>> Regards, Igor Goncharovsky
>> Unistim Dev: http://unistim.igorg.ru
>>
>>
>> --
>> _
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>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
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>>
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>>
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>>
>
>
>
> --
> George Joseph
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
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>
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>



-- 
Regards, Igor Goncharovsky
Unistim Dev: http://unistim.igorg.ru
Blog: http://igorg.ru
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[asterisk-users] Asterisk 11 - Security Fix Only Notice

2016-10-25 Thread Matt Fredrickson 
Hey All,

This is a friendly notice that as of today Asterisk 11 has entered
security fix only mode.  From this point onward additional releases of
Asterisk 11 will no longer be made unless there is a security fix
being applied to the branch.  Users of Asterisk 11 are encouraged to
move to one of the newer major versions, Asterisk 13 (LTS) or Asterisk
14, as soon as possible.

For more information on Asterisk versions and their supported lifetimes,
please see the following wiki page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Thank you for your continued support of Asterisk!

-- 
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] Asterisk 14.1.0 Now Available

2016-10-25 Thread Asterisk Development Team 
The Asterisk Development Team has announced the release of Asterisk 14.1.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.1.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
---
 * ASTERISK-26277 - Add dialplan function
  PJSIP_SEND_SESSION_REFRESH that sends a session refresh to
  update formats on a channel after session establishment
  (Reported by Matt Jordan)

Bugs fixed in this release:
---
 * ASTERISK-26477 - pjproject:  SEGV during SSL operations
  (Reported by George Joseph)
 * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk 
  allows one end peer to send video, even though the other end
  supports only audio. (Reported by effie mouzeli)
 * ASTERISK-26416 - pjproject-bundled: configure fails to check for
  all required utilities (Reported by Corey Farrell)
 * ASTERISK-26466 - core: Be forgiving on external callerid that
  may be flawed so we don't drop events (Reported by Richard
  Mudgett)
 * ASTERISK-26362 - res_config_mysql:  Broken after 13.10 (Reported
  by Carlos Chavez)
 * ASTERISK-26446 - app_dial:  There's no way to override the
  hangupcause on unanswered channels (Reported by George Joseph)
 * ASTERISK-26410 - core: Asterisk 14 doesn't show the header in
  the console or verbose when starting (Reported by Dan Jenkins)
 * ASTERISK-24311 - Populating database via Alembic fails when
  using same database for multiple schema sets (Reported by Dafi
  Ni)
 * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
  Symmetric Response. (Reported by Alexander Traud)
 * ASTERISK-26426 - format_ogg_opus: remove from source (Reported
  by Kevin Harwell)
 * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
  bindaddr=[::] (Reported by Jacek)
 * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows
  do_monitor lock (Reported by Barry Flanagan)
 * ASTERISK-26397 - manager: PresenceState action crashes Asterisk
  14 (Reported by Andrew Nagy)
 * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by
  Joshua Colp)
 * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
  instructed (Reported by Tzafrir Cohen)
 * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled
  (Reported by Anthony Messina)
 * ASTERISK-26391 - Consoles do not display verbose logger messages
  even when requested. (Reported by Marcelo Terres)
 * ASTERISK-26263 - SQL error when using realtime and registering
  extension / inserting into ps_contacts (Reported by Jeppe Ryskov
  Larsen)
 * ASTERISK-26365 - rtp: Offer with multiple payloads for same
  codec is incorrectly handled (Reported by Joshua Colp)
 * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
  by Aaron Hamstra)
 * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten
  for connectionful protocols (Reported by Joshua Colp)
 * ASTERISK-26367 - rtp: Timestamps broken when video frame is
  across multiple RTP packets (Reported by Joshua Colp)
 * ASTERISK-26375 - res_pjsip_transport_management: Log message
  states seconds, but time value is milliseconds (Reported by
  Joshua Colp)
 * ASTERISK-26364 - res_pjsip: Don't assume a request will have
  target addresses (Reported by Joshua Colp)
 * ASTERISK-26360 - app_queue: "queue show" output gets "failed to
  extend from 240 to 327" msgs. (Reported by Richard Mudgett)
 * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes
  unexpected callerid (Reported by Kevin Harwell)
 * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
  'REGISTER' failed (Reported by Dmitry Melekhov)
 * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
  (Reported by Etienne Lessard)
 * ASTERISK-26264 - res_pjsip: Crash when applying ACL from
  non-existent endpoint (Reported by nappsoft)
 * ASTERISK-26341 - ARI: Stopping a media playlist only stops the
  current media URI being played back, and not the whole list
  (Reported by Matt Jordan)
 * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p'
  argument) is enabled and callee rejects a call or hangs up.
  (Reported by Etienne Lessard)
 * ASTERISK-26331 - Crash on “core show channeltype Surrogate” in
  ast_format_cap_get_names (Reported by CGI.NET)
 * ASTERISK-26269 - res_pjsip: Wrong state for aors without
  registered contacts after startup (Reported by nappsoft)
 * ASTERISK-26282 - AEL: macro-call in Dial application, macro
  "lacks 's' extension" (Reported by chris de rock)
 * ASTERISK-26226 - pbx: Asterisk crash on AMI action
  "ShowDialplan" when there's a circular dependency

[asterisk-users] Asterisk 13.12.0 Now Available

2016-10-25 Thread Asterisk Development Team 
The Asterisk Development Team has announced the release of Asterisk 13.12.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.12.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
---
 * ASTERISK-26277 - Add dialplan function
  PJSIP_SEND_SESSION_REFRESH that sends a session refresh to
  update formats on a channel after session establishment
  (Reported by Matt Jordan)

Bugs fixed in this release:
---
 * ASTERISK-26477 - pjproject:  SEGV during SSL operations
  (Reported by George Joseph)
 * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk 
  allows one end peer to send video, even though the other end
  supports only audio. (Reported by effie mouzeli)
 * ASTERISK-26416 - pjproject-bundled: configure fails to check for
  all required utilities (Reported by Corey Farrell)
 * ASTERISK-26466 - core: Be forgiving on external callerid that
  may be flawed so we don't drop events (Reported by Richard
  Mudgett)
 * ASTERISK-26362 - res_config_mysql:  Broken after 13.10 (Reported
  by Carlos Chavez)
 * ASTERISK-26446 - app_dial:  There's no way to override the
  hangupcause on unanswered channels (Reported by George Joseph)
 * ASTERISK-24311 - Populating database via Alembic fails when
  using same database for multiple schema sets (Reported by Dafi
  Ni)
 * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
  Symmetric Response. (Reported by Alexander Traud)
 * ASTERISK-26426 - format_ogg_opus: remove from source (Reported
  by Kevin Harwell)
 * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
  bindaddr=[::] (Reported by Jacek)
 * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows
  do_monitor lock (Reported by Barry Flanagan)
 * ASTERISK-26397 - manager: PresenceState action crashes Asterisk
  14 (Reported by Andrew Nagy)
 * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by
  Joshua Colp)
 * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
  instructed (Reported by Tzafrir Cohen)
 * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled
  (Reported by Anthony Messina)
 * ASTERISK-26263 - SQL error when using realtime and registering
  extension / inserting into ps_contacts (Reported by Jeppe Ryskov
  Larsen)
 * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten
  for connectionful protocols (Reported by Joshua Colp)
 * ASTERISK-26367 - rtp: Timestamps broken when video frame is
  across multiple RTP packets (Reported by Joshua Colp)
 * ASTERISK-26375 - res_pjsip_transport_management: Log message
  states seconds, but time value is milliseconds (Reported by
  Joshua Colp)
 * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
  by Aaron Hamstra)
 * ASTERISK-26360 - app_queue: "queue show" output gets "failed to
  extend from 240 to 327" msgs. (Reported by Richard Mudgett)
 * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes
  unexpected callerid (Reported by Kevin Harwell)
 * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
  'REGISTER' failed (Reported by Dmitry Melekhov)
 * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
  (Reported by Etienne Lessard)
 * ASTERISK-26264 - res_pjsip: Crash when applying ACL from
  non-existent endpoint (Reported by nappsoft)
 * ASTERISK-26288 - followme: fails to reset config items to
  default values on reload (Reported by Tzafrir Cohen)
 * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p'
  argument) is enabled and callee rejects a call or hangs up.
  (Reported by Etienne Lessard)
 * ASTERISK-26331 - Crash on “core show channeltype Surrogate” in
  ast_format_cap_get_names (Reported by CGI.NET)
 * ASTERISK-26282 - AEL: macro-call in Dial application, macro
  "lacks 's' extension" (Reported by chris de rock)
 * ASTERISK-26226 - pbx: Asterisk crash on AMI action
  "ShowDialplan" when there's a circular dependency between
  contexts (Reported by Etienne Lessard)
 * ASTERISK-26279 - pjproject-bundled:  Fails to compile on Debian
  6 (Reported by George Joseph)
 * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
  cleaning up properly (Reported by Alexander Traud)
 * ASTERISK-26299 - app_queue: Queue application sometimes stops
  calling members with Local interface (Reported by Etienne
  Lessard)
 * ASTERISK-26203 - res_fax: Deadlock when using
  FAXOPT(gateway)=yes with Local channels (Reported by Etienne
  Lessard)
 * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking
  inversion in T.38 query option with

[asterisk-users] Asterisk 11.24.0 Now Available

2016-10-25 Thread Asterisk Development Team 
The Asterisk Development Team has announced the release of Asterisk 11.24.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.24.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
  Symmetric Response. (Reported by Alexander Traud)
 * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
  bindaddr=[::] (Reported by Jacek)
 * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
  instructed (Reported by Tzafrir Cohen)
 * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
  by Aaron Hamstra)
 * ASTERISK-26360 - app_queue: "queue show" output gets "failed to
  extend from 240 to 327" msgs. (Reported by Richard Mudgett)
 * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
  (Reported by Etienne Lessard)
 * ASTERISK-26288 - followme: fails to reset config items to
  default values on reload (Reported by Tzafrir Cohen)
 * ASTERISK-26282 - AEL: macro-call in Dial application, macro
  "lacks 's' extension" (Reported by chris de rock)
 * ASTERISK-26226 - pbx: Asterisk crash on AMI action
  "ShowDialplan" when there's a circular dependency between
  contexts (Reported by Etienne Lessard)
 * ASTERISK-26299 - app_queue: Queue application sometimes stops
  calling members with Local interface (Reported by Etienne
  Lessard)
 * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
  cleaning up properly (Reported by Alexander Traud)
 * ASTERISK-26203 - res_fax: Deadlock when using
  FAXOPT(gateway)=yes with Local channels (Reported by Etienne
  Lessard)
 * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking
  inversion in T.38 query option with features bridging code
  (Reported by David Brillert)
 * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
  local channels. (Reported by Richard Mudgett)
 * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when
  channels have multiple native formats (Reported by Matt Jordan)
 * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
  SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
  abelbeck)
 * ASTERISK-25706 - pbx: Abort asterisk on features reload
  (handle_hint_change) (Reported by Krzysztof Trempala)
 * ASTERISK-26233 - pbx: Failure to remove inconsistent extension
  names (Reported by Corey Farrell)
 * ASTERISK-26267 - ast_register_atexit callbacks should be run on
  failed startup. (Reported by Corey Farrell)
 * ASTERISK-26265 - Errors ignored from some parts of system
  initialization. (Reported by Corey Farrell)
 * ASTERISK-25996 - Remove "live_dangerously" requirement on
  DB(read) (Reported by Andrew Nagy)
 * ASTERISK-26237 - Fax is detected on regular calls. (Reported by
  Richard Mudgett)
 * ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
  command and attended transfer handling (Reported by Ben
  Smithurst)
 * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in
  conditional code. (Reported by Corey Farrell)
 * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence
  number even on lost packets. (Reported by Alexander Traud)
 * ASTERISK-26038 - 'make install' doesn't seem to install OS/X
  init files (Reported by Tzafrir Cohen)
 * ASTERISK-26133 - app_queue: Queue members receive multiple calls
  (Reported by Richard Miller)
 * ASTERISK-26196 - pbx: Time based includes can leak timezone
  string (Reported by Corey Farrell)
 * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing
  DTLS failure occurred on RTP instance (Reported by Edwin
  Vandamme)
 * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
  (Reported by Alexander Traud)
 * ASTERISK-25289 - Build System does not respect CFLAGS and
  CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
 * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out
  of bounds and bugs (Reported by Alexei Gradinari)
 * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
  by Joshua Colp)
 * ASTERISK-26157 - Build:   Fix errors highlighted by GCC 6.x
  (Reported by George Joseph)

Improvements made in this release:
---
 * ASTERISK-26220 - Add support for noreturn function attributes.
  (Reported by Corey Farrell)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0

Thank you for your continued support of Asterisk!

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[asterisk-users] AST-2016-007: UPDATE

2016-10-25 Thread Asterisk Security Team 
On September 8, the Asterisk development team released the AST-2016-007
security advisory. The security advisory involved an RTP resource
exhaustion that could be targeted due to a flaw in the "allowoverlap"
option of chan_sip. Due to new information presented to us by Walter
Doekes, we have made the following updates to the advisory.

In the "Description" section, the following text has been added:

UPDATE (20 October, 2016):   

  
It has been brought to our attention by Walter Doekes that
this same leak can be exploited without the use of the
overlap dialing feature. Sending SIP requests in a specific
sequence outside the norm could also cause the leak of RTP
resources. By sending an in-dialog INVITE after receiving a
404 response (but before sending an ACK), an attacker could
cause the same leak to occur."

In the "Resolution" section, the following text has been added:

UPDATE (20 October, 2016):
  
Because of the updated information from Walter Doekes,
disabling the allowoverlap option is not enough to solve
this issue. Users of Asterisk MUST upgrade to one of the
fixed versions listed below.

The updated advisory can be found at
http://downloads.asterisk.org/pub/security/AST-2016-007.html
and
http://downloads.asterisk.org/pub/security/AST-2016-007.pdf


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Re: [asterisk-users] Opus codec in codecs.conf

2016-10-25 Thread Joshua Colp 

j run wrote:

any git tag in particular? (13.12.0-rc1 is my best guess)


Yes, that is the current release candidate.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


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Re: [asterisk-users] Opus codec in codecs.conf

2016-10-25 Thread j run 
any git tag in particular? (13.12.0-rc1 is my best guess)

jrun
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Re: [asterisk-users] Opus codec in codecs.conf

2016-10-25 Thread jrun 
any updates as to when this would be available for 13?


jrun

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Re: [asterisk-users] Opus codec in codecs.conf

2016-10-25 Thread Joshua Colp 

jrun wrote:

any updates as to when this would be available for 13?


The version of 13 with the changes required to support it is currently 
in release candidate status. I expect it to reach release status soon.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


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Re: [asterisk-users] Opus codec in codecs.conf

2016-10-25 Thread George Joseph 
On Mon, Oct 24, 2016 at 6:54 PM, Igor Goncharovsky <
igor.goncharov...@gmail.com> wrote:

> Hello,
>
> I am trying to configure new opus codec in asterisk 14, but unable to find
> any examples of codecs.conf settings for this codec.
>
> All I am trying to do - setup peer with using opus in narrow band mode
> (8kHz sampling rate). Does anybody know how to configure chan_opus?
>
>
If you run "config show help condec_opus opus" from teh Asterisk command
line, you'll get a list of the configuration options

pbx1*CLI> config show help codec_opus opus
opus: [category !~ /.?/]

Codec opus module for Asterisk options

type  -- Must be of type 'opus'
sample_rate   -- Codec's sample rate.
packet_loss   -- Encoder's packet loss percentage.

complexity-- Encoder's computational complexity.

max_bandwidth -- Encoder's maximum bandwidth allowed.

signal-- Encoder's signal type.
application   -- Encoder's application type.
max_playback_rate -- Encoder's maximum playback rate.

max_ptime -- Encoder's maximum packetization rate.

ptime -- Encoder's packetization rate.
bitrate   -- Encoder's bit rate.
cbr   -- Encoder's constant bit rate value.
fec   -- Encoder's forward error correction value.
dtx   -- Encoder's discontinuous transmission value.



> --
> Regards, Igor Goncharovsky
> Unistim Dev: http://unistim.igorg.ru
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users