Re: [asterisk-users] Asterisk 13.19.0 Now Available

2018-01-11 Thread Binarus 
On 11.01.2018 20:51, Asterisk Development Team wrote:
> The Asterisk Development Team would like to announce the release of
> Asterisk 13.19.0.
> This release is available for immediate download at
> http://downloads.asterisk.org/pub/telephony/asterisk
> 
> The release of Asterisk 13.19.0 resolves several issues reported by the
> community and would have not been possible without your participation.
> 
> *Thank you!*

Thank you very much for caring so much about security and bug fixes!

But in this case, I am slightly worried. I saw the announcements for
version 13 and version 15, but no announcement for version 14 yet. The
website currently still offers 14.7.5 for download.

Could this be one of the rare cases where 13 and 15 needed security
fixes, but 14 didn't?

Thank you very much,

Binarus

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Re: [asterisk-users] Asterisk 15.2.0 Now Available

2018-01-11 Thread Joshua Colp 
On Thu, Jan 11, 2018, at 4:34 PM, Ira wrote:
> Re: [asterisk-users] Asterisk 15.2.0 Now AvailableHello Asterisk,
> 
>  Thursday, January 11, 2018, 11:59:28 AM, you wrote:
> 
> 
> The release of Asterisk 15.2.0 resolves several issues reported by the
> community and would have not been possible without your participation.
> For amusement I tried this again. I'm running an old version of 32 bit
> CentOS, one that just went off support and when I run ./configure it
> fails while trying to download pjproject. It looks like it says it's
> going to try, fails, tries again and then gives up. Running with
> NOISY_BUILD does not any useful information. Looks like it never tries
> to execute the download command unless it executes it silently.
> 
>  It's not important, I'm perfectly happy running 14, but always try to
>  run the most current if I can.

Can you please file an issue[1] with all the information?

[1] https://issues.asterisk.org/jira

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Re: [asterisk-users] Asterisk 15.2.0 Now Available

2018-01-11 Thread Ira 
Title: Re: [asterisk-users] Asterisk 15.2.0 Now Available


Hello Asterisk,

Thursday, January 11, 2018, 11:59:28 AM, you wrote:





The release of Asterisk 15.2.0 resolves several issues reported by the
community and would have not been possible without your participation.



For amusement I tried this again. I'm running an old version of 32 bit CentOS, one that just went off support and when I run ./configure it fails while trying to download pjproject. It looks like it says it's going to try, fails, tries again and then gives up. Running with NOISY_BUILD does not any useful information. Looks like it never tries to execute the download command unless it executes it silently.

It's not important, I'm perfectly happy running 14, but always try to run the most current if I can.

-- Ira


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[asterisk-users] Asterisk 15.2.0 Now Available

2018-01-11 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
15.2.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 15.2.0 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:

New Features made in this release:
---
 * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
  incoming INVITE Request-URI.
  (Reported by Richard Mudgett)
 * ASTERISK-27413 - Add cache_media_frames debugging option.
   
  (Reported by Richard Mudgett)
 * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
  endpoint identification to IP only
  (Reported by Ben
  Merrills)

Bugs fixed in this release:
---
 * ASTERISK-27531 - Compiler optimizations can break module load
  sequence.
  (Reported by abelbeck)
 * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
  Contact crashes asterisk
  (Reported by Ross Beer)
 * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
  read()
  (Reported by Abhay Gupta)
 * ASTERISK-25079 - AMI bridge of channels results in MOH not
  destroyed and robotic audio on one channel
  (Reported by
  Zane Conkle)
 * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash

  (Reported by Corey Farrell)
 * ASTERISK-27490 - chan_console: 'set active' fails to work
   
  (Reported by Tzafrir Cohen)
 * ASTERISK-24756 - ConfBridge sound_muted does not work from
  CLI or AMI
  (Reported by Thomas Frederiksen)
 * ASTERISK-25649 - Transfer application does not work with
  Local channels - documentation misleading
  (Reported by
  Ivan Ullmann)
 * ASTERISK-25869 - chan_sip: "rejected because extension not
  found" should be logged as a security event
  (Reported by
  Brian J. Murrell)
 * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
  streams
  (Reported by Wim De Vlaminck)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
  music the first time it is played
  (Reported by Thomas
  Frederiksen)
 * ASTERISK-19657 - Coverity Report: Fix issues for error type
  CHAR_IO
  (Reported by Matt Jordan)
 * ASTERISK-27175 - iax.conf demo peer is invalid
 
  (Reported by Tzafrir Cohen)
 * ASTERISK-27430 - README refers to security documents that do
  not exist.
  (Reported by Corey Farrell)
 * ASTERISK-20281 - "core set verbose" behaves strangely, can't
  alias it, cli.conf example broken
  (Reported by Tim
  Ringenbach at Asteria Solutions Group)
 * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
  FXS gateway
  (Reported by Tzafrir Cohen)
 * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
  RTCP packet will write past where it should
  (Reported by
  Vitezslav Novy)
 * ASTERISK-27408 - Identify causes and fix
  pjsip/resolver/srv/failover/in_dialog/transport_tcp
 
  (Reported by Corey Farrell)
 * ASTERISK-18411 - Queue members with hints for state_interface
  get stuck in "In Use" state.
  (Reported by Steven T.
  Wheeler)
 * ASTERISK-26131 - chan_sip: Crash Asterisk (in
  sip_request_call at chan_sip.c) by making a call to a single
  character in a dot pattern match
  (Reported by Dwayne
  Hubbard)
 * ASTERISK-27475 - codec_opus requires libcurl
  (Reported
  by Samuel For)
 * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
  not applied on reload
  (Reported by John Bigelow)
 * ASTERISK-27465 - CLI Completion Not Working
  (Reported
  by Ross Beer)
 * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
  Variable CDR(amaflags)=...
  (Reported by Richard Mudgett)
 * ASTERISK-27453 - RTP: Blind transfer direct media scenario
  results in one way audio.
  (Reported by Richard Mudgett)
 * ASTERISK-20643 - SIP ICE support - remove hardcoded
  limitation on SDP size, make ICE support disabled by default in
  SIP, maybe provide a better warning message
  (Reported by
  Roy)
 * ASTERISK-26980 - pjsip: Clean up WebRTC disables
 
  (Reported by abelbeck)
 * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
  flooded with unauthenticated requests
  (Reported by George
  Joseph)
 * ASTERISK-27454 - res_http_post: Don't require
  GMIME_MAJOR_VERSION
  (Reported by Joshua Colp)
 * ASTERISK-23735 - Transcoding makes bad choice in high-rate
  translations
  (Reported by Richard Kenner)
 * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
  message.
  (Reported by Frank Durden)
 * ASTERISK-24662 - [patch] column and row headers for Signed
  Linear format variants in output of 'core show translation' are
  ambiguous

[asterisk-users] Asterisk 13.19.0 Now Available

2018-01-11 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
13.19.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.19.0 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:

New Features made in this release:
---
 * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
  incoming INVITE Request-URI.
  (Reported by Richard Mudgett)
 * ASTERISK-27413 - Add cache_media_frames debugging option.
   
  (Reported by Richard Mudgett)
 * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
  endpoint identification to IP only
  (Reported by Ben
  Merrills)

Bugs fixed in this release:
---
 * ASTERISK-27531 - Compiler optimizations can break module load
  sequence.
  (Reported by abelbeck)
 * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
  Contact crashes asterisk
  (Reported by Ross Beer)
 * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
  read()
  (Reported by Abhay Gupta)
 * ASTERISK-25079 - AMI bridge of channels results in MOH not
  destroyed and robotic audio on one channel
  (Reported by
  Zane Conkle)
 * ASTERISK-27490 - chan_console: 'set active' fails to work
   
  (Reported by Tzafrir Cohen)
 * ASTERISK-24756 - ConfBridge sound_muted does not work from
  CLI or AMI
  (Reported by Thomas Frederiksen)
 * ASTERISK-25649 - Transfer application does not work with
  Local channels - documentation misleading
  (Reported by
  Ivan Ullmann)
 * ASTERISK-25869 - chan_sip: "rejected because extension not
  found" should be logged as a security event
  (Reported by
  Brian J. Murrell)
 * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
  streams
  (Reported by Wim De Vlaminck)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
  music the first time it is played
  (Reported by Thomas
  Frederiksen)
 * ASTERISK-19657 - Coverity Report: Fix issues for error type
  CHAR_IO
  (Reported by Matt Jordan)
 * ASTERISK-27175 - iax.conf demo peer is invalid
 
  (Reported by Tzafrir Cohen)
 * ASTERISK-27430 - README refers to security documents that do
  not exist.
  (Reported by Corey Farrell)
 * ASTERISK-20281 - "core set verbose" behaves strangely, can't
  alias it, cli.conf example broken
  (Reported by Tim
  Ringenbach at Asteria Solutions Group)
 * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
  FXS gateway
  (Reported by Tzafrir Cohen)
 * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
  RTCP packet will write past where it should
  (Reported by
  Vitezslav Novy)
 * ASTERISK-27408 - Identify causes and fix
  pjsip/resolver/srv/failover/in_dialog/transport_tcp
 
  (Reported by Corey Farrell)
 * ASTERISK-18411 - Queue members with hints for state_interface
  get stuck in "In Use" state.
  (Reported by Steven T.
  Wheeler)
 * ASTERISK-26131 - chan_sip: Crash Asterisk (in
  sip_request_call at chan_sip.c) by making a call to a single
  character in a dot pattern match
  (Reported by Dwayne
  Hubbard)
 * ASTERISK-27475 - codec_opus requires libcurl
  (Reported
  by Samuel For)
 * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
  not applied on reload
  (Reported by John Bigelow)
 * ASTERISK-27465 - CLI Completion Not Working
  (Reported
  by Ross Beer)
 * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
  Variable CDR(amaflags)=...
  (Reported by Richard Mudgett)
 * ASTERISK-27453 - RTP: Blind transfer direct media scenario
  results in one way audio.
  (Reported by Richard Mudgett)
 * ASTERISK-20643 - SIP ICE support - remove hardcoded
  limitation on SDP size, make ICE support disabled by default in
  SIP, maybe provide a better warning message
  (Reported by
  Roy)
 * ASTERISK-26980 - pjsip: Clean up WebRTC disables
 
  (Reported by abelbeck)
 * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
  flooded with unauthenticated requests
  (Reported by George
  Joseph)
 * ASTERISK-27454 - res_http_post: Don't require
  GMIME_MAJOR_VERSION
  (Reported by Joshua Colp)
 * ASTERISK-23735 - Transcoding makes bad choice in high-rate
  translations
  (Reported by Richard Kenner)
 * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
  message.
  (Reported by Frank Durden)
 * ASTERISK-24662 - [patch] column and row headers for Signed
  Linear format variants in output of 'core show translation' are
  ambiguous
  (Reported by Rusty Newton)
 * ASTERISK-27353 - H323 audio starts with a delay of 2
  seconds.

[asterisk-users] Is Asterisk 11's chan_sip able to send RTCP reports ?

2018-01-11 Thread Olivier 
Hello,

On a lab setup, I can see an Asterisk 11 system is correctly receiving and
displaying (sip show channelstats) incoming RTCP reports but not any report
to the other end.

Searching through *.sample files does show much.
This highlighted I still have a lot to learn on RTCP.

My setup is:
Asterisk 13  <---> Asterisk 11

In my tests, Asterisk 13 box calls Asterisk 11 box.


1. In Asterisk 11 box, there is no RTCP trace in incoming INVITE's SDP as
if RTCP is implied. RFC3605 mentions rtcp attributes but also implicit
communication. Is it common practice for SIP devices to send RTCP reports
on implicit ports (ie port computed from RTP port) without explicit mention
in INVITE message or more generally without negotiating with the other end ?

2. Is there a way to toggle on or off RTCP sending in Asterisk for all SIP
peers or PJSIP endpoints ?

3. Is there a way to toggle on or off RTCP sending in Asterisk for a given
SIP peer or PJSIP endpoint ?

4. Same as 2 and 3 but with RTCP receiving (ie dumping incoming RTCP
messages) ?

5. Am I correct in thinking RTCP stats are meaningful with previous or next
hop ? In other words, if a communication chain such as A <--> B <--> C <-->
D, packet lost between A and B are not reported to C.

6. Does RTPProxy support RTCP XR stats ?

Best regards
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Re: [asterisk-users] Logging ARI debug messages

2018-01-11 Thread Floimair Florian 
Thanks for the quick reply Joshua!

I might dig into this and try an implementation.

 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Joshua Colp
Gesendet: Donnerstag, 11. Januar 2018 16:35
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Logging ARI debug messages

On Thu, Jan 11, 2018, at 11:30 AM, Floimair Florian wrote:
> Hi there!
> 
> Is there any way I can turn on debug for ARI and sending the output to 
> a separate log file?
> So far I have only been able to turn on ARI debugging in the console 
> which results in the debug output being logged in /var/log/asterisk/ 
> messages
> 
> I would love to have ARI debug log messages in /var/log/asterisk/debug 
> or even better in it's own ari-debug file.

That is not something anyone has implemented as of this time. The messages 
themselves just get raised as normal verbose messages.

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

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Re: [asterisk-users] Logging ARI debug messages

2018-01-11 Thread Joshua Colp 
On Thu, Jan 11, 2018, at 11:30 AM, Floimair Florian wrote:
> Hi there!
> 
> Is there any way I can turn on debug for ARI and sending the output to a 
> separate log file?
> So far I have only been able to turn on ARI debugging in the console 
> which results in the debug output being logged in /var/log/asterisk/
> messages
> 
> I would love to have ARI debug log messages in /var/log/asterisk/debug 
> or even better in it's own ari-debug file.

That is not something anyone has implemented as of this time. The messages 
themselves just get raised as normal verbose messages.

-- 
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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[asterisk-users] Logging ARI debug messages

2018-01-11 Thread Floimair Florian 
Hi there!

Is there any way I can turn on debug for ARI and sending the output to a 
separate log file?
So far I have only been able to turn on ARI debugging in the console which 
results in the debug output being logged in /var/log/asterisk/messages

I would love to have ARI debug log messages in /var/log/asterisk/debug or even 
better in it's own ari-debug file.



With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

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Re: [asterisk-users] disable call features

2018-01-11 Thread Kseniya Blashchuk 
Ah seems I can just unload res_features.so ))

On Thu, Jan 11, 2018, 4:56 PM Kseniya Blashchuk  wrote:

> Hi all!
> Does anybody know if it's possible to completely disable all asterisk call
> features (even the default ones like xfer)?
> Thanks in advance
>
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[asterisk-users] disable call features

2018-01-11 Thread Kseniya Blashchuk 
Hi all!
Does anybody know if it's possible to completely disable all asterisk call
features (even the default ones like xfer)?
Thanks in advance
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