Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

2021-02-18 Thread Luca Bertoncello
Am 18.02.2021 um 18:59 schrieb Michael Maier:
> On 17.02.21 at 21:46 Luca Bertoncello wrote:
>> Am 16.02.2021 um 22:32 schrieb Michael Maier:
>>
>> Hi Michael
>>
 Maybe could you send me an abstract of your configuration?
>>>
>>> Take a look here [1]
>>
>> So, maybe I got it...
>> I tested the configuration with my Fax number and it seems to work (= I
>> can call the fax and can call my mobile phone from the fax with
>> "originate...").
> 
> Congrats!

So, it seems it does NOT work as expected...
I tried to activate the FAX and it works, then I activated my number and
it works, too.
Finally I activated the number of my wife and it does not work anymore...
If I call the number I can only see (verbose 42):

[Feb 18 19:57:12] NOTICE[19379] res_pjsip/pjsip_distributor.c: Request
'INVITE' from ''
failed for '217.0.21.64:5060' (callid: p65550t1613674632m753568c93349s2)
- No matching endpoint found

and no phone rings...
After that, even if I restore the single number to SIP I only get the
error and nothing work, until I restored _ALL_ numbers to SIP.

Do someone has an explanation and (better!) a solution to the problem?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

2021-02-18 Thread Michael Maier
On 17.02.21 at 21:46 Luca Bertoncello wrote:
> Am 16.02.2021 um 22:32 schrieb Michael Maier:
> 
> Hi Michael
> 
>>> Maybe could you send me an abstract of your configuration?
>>
>> Take a look here [1]
> 
> So, maybe I got it...
> I tested the configuration with my Fax number and it seems to work (= I
> can call the fax and can call my mobile phone from the fax with
> "originate...").

Congrats!

> On the registration I have:
> 
> [pbxfax]
> type = registration
> retry_interval = 20
> max_retries = 10
> contact_user = 00493514977291
> expiration = 120
> transport = transport-udp
> outbound_auth = pbxfax
> client_uri = sip:03514977...@tel.t-online.de
> server_uri = sip:tel.t-online.de
> 
> First: can I use tel.t-online.de or _MUST_ I change it?

No, you mustn't change it. You must use tel.t-online.de.

> If I understand
> your previous E-Mail, I'd say that I can leave tel.t-online.de...

Correctly!

> Then I have a question by the Dialplan... Currently I have:
> 
> [fax-out]
> exten => _X.,1,NoOp()
> exten => _X.,n,Verbose(2,Call from FAX)
> exten => _X.,n,Dial(SIP/pbxfax/${EXTEN},,R)
> 
> And I'll replace it with:
> 
> [fax-out]
> exten => _X.,1,NoOp()
> exten => _X.,n,Verbose(2,Call from FAX)
> exten => _X.,n,Dial(PJSIP/pbxfax/sip:${EXTEN}@tel.t-online.de,,R)
> 
> Is it correct? I tried with
> "PJSIP/pbxfax/pjsip:${EXTEN}@tel.t-online.de,,R" and it does NOT work...
> Is it correct, that I have to leave "sip:..."?

Don't know - I don't care about dialplan - I'm using FreePBX :-)


Thanks
Michael

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[asterisk-users] AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver

2021-02-18 Thread Asterisk Security Team
   Asterisk Project Security Advisory - AST-2021-005

  ProductAsterisk 
  SummaryRemote Crash Vulnerability in PJSIP channel driver   
Nature of Advisory   Denial of Service
  Susceptibility Remote Unauthenticated Sessions  
 SeverityModerate 
  Exploits Known No   
Reported On  December 4, 2020 
Reported By  Mauri de Souza Meneguzzo (3CPlus)
 Posted On   February 8, 2021 
  Last Updated OnFebruary 8, 2021 
 Advisory ContactJcolp AT sangoma DOT com 
 CVE NameCVE-2021-26906   

  Description Given a scenario where an outgoing call is placed from  
  Asterisk to a remote SIP server it is possible for a
  crash to occur. 
  
  The code responsible for negotiating SDP in SIP 
  responses incorrectly assumes that SDP negotiation  
  will always be successful. If a SIP response
  containing an SDP that can not be negotiated is 
  received a subsequent SDP negotiation on the same call  
  can cause a crash.  
  
  If the “accept_multiple_sdp_answers” option in the
  
  “system” section of pjsip.conf is set to “yes” 
then 
  any subsequent non-forked SIP response with SDP can 
  trigger this crash. 
  
  If the “follow_early_media_fork” option in the
  
  “system” section of pjsip.conf is set to “yes” 
(the 
  default) then any subsequent SIP responses with SDP 
  from a forked destination can trigger this crash.   
  
  If a 200 OK with SDP is received from a forked  
  destination it can also trigger this crash, even if 
  the “follow_early_media_fork” option is not set to
  
  “yes”.
  
  
  In all cases this relies on a race condition with   
  tight timing where the second SDP negotiation occurs
  before termination of the call due to the initial SDP   
  negotiation failure.
Modules Affected  res_pjsip_session.c, PJSIP  

Resolution  The issue has been fixed in PJSIP by changing the behavior
of the pjmedia_sdp_neg_modify_local_offer2 function. If SDP   
was previously negotiated the code no longer assumes that it  
was successful and instead checks that SDP was negotiated.
  
This issue can only be resolved by upgrading to a fixed   
version or applying the provided patch.   

   Affected Versions
Product  Release Series  
 Asterisk Open Source 13.x   All versions 
 Asterisk Open Source 16.x   All versions 
 Asterisk Open Source 17.x   All versions 
 Asterisk Open Source 18.x   All versions 
  Certified Asterisk  16.x   All versions 

  Corrected In
   Product  Release   
Asterisk Open Source   13.38.2, 16.16.1, 17.9.2, 18.2.1   
 Certified Asterisk   16.8-cert6  

 Patches  
   Patch URL  Revision  
   htt

[asterisk-users] AST-2021-004: An unsuspecting user could crash Asterisk with multiple hold/unhold requests

2021-02-18 Thread Asterisk Security Team
   Asterisk Project Security Advisory - AST-2021-004

 ProductAsterisk  
 SummaryAn unsuspecting user could crash Asterisk with
multiple hold/unhold requests 
Nature of Advisory  Denial of Service 
  SusceptibilityRemote authenticated sessions 
 Severity   Moderate  
  Exploits KnownNo
   Reported On  December 9, 2020  
   Reported By  Edvin Vidmar  
Posted On   
 Last Updated OnFebruary 11, 2021 
 Advisory Contact   gjoseph AT sangoma DOT com
 CVE Name   CVE-2021-26714

  Description Due to a signedness comparison mismatch, an 
  authenticated WebRTC client could cause a stack 
  overflow and Asterisk crash by sending multiple 
  hold/unhold requests in quick succession.   
Modules Affected  res_rtp_asterisk.c  

  ResolutionThe packet size comparison terms have been corrected. 

   Affected Versions
Product   Release Series  
  Asterisk Open Source 16.x   16.16.0 
  Asterisk Open Source 17.x   17.9.1  
  Asterisk Open Source 18.x   18.2.0  
   Certified Asterisk  16.x   16.8-cert5  

  Corrected In
 Product  Release 
   Asterisk Open Source   16.16.1, 17.9.2, 18.2.1 
Certified Asterisk   16.8-cert6   

 Patches 
   Patch URL  Revision  
   https:/downloads.asterisk.org/pub/security/AST-2021-004-16.diff   Asterisk   
 16 
   https:/downloads.asterisk.org/pub/security/AST-2021-004-17.diff   Asterisk   
 17 
   https:/downloads.asterisk.org/pub/security/AST-2021-004-18.diff   Asterisk   
 18 
   https:/downloads.asterisk.org/pub/security/AST-2021-004-16.8.diff Certified  
 Asterisk   
 16.8-cert6 

 Links   https://issues.asterisk.org/jira/browse/ASTERISK-29205   
  
 https://downloads.asterisk.org/pub/security/AST-2021-004.html

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security  
  
This document may be superseded by later versions; if so, the latest  
version will be posted at 
https://downloads.digium.com/pub/security/AST-2021-004.pdf and
https://downloads.digium.com/pub/security/AST-2021-004.html   

Revision History
  Date  Editor Revisions Made 
February 4, 2021   George Joseph Initial revision 
February 9, 2021   George Joseph Added CVE

   Asterisk Project Security Advisory - AST-2021-004
   Copyright © 2021 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
   original, unaltered form.

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[asterisk-users] AST-2021-003: Remote attacker could prematurely tear down SRTP calls

2021-02-18 Thread Asterisk Security Team
   Asterisk Project Security Advisory - AST-2021-003

 ProductAsterisk  
 SummaryRemote attacker could prematurely tear down SRTP  
calls 
Nature of Advisory  Denial of Service 
  SusceptibilityRemote unauthenticated sessions   
 Severity   Moderate  
  Exploits KnownNo
   Reported On  January 22, 2021  
   Reported By  Alexander Traud   
Posted On   
 Last Updated OnFebruary 11, 2021 
 Advisory Contact   gjoseph AT sangoma DOT com
 CVE Name   CVE-2021-26712

  Description An unauthenticated remote attacker could replay SRTP
  packets which could cause an Asterisk instance  
  configured without strict RTP validation to tear down   
  calls prematurely.  
Modules Affected  res_srtp.c res_rtp_asterisk.c   

Resolution  Asterisk now implements SRTP replay protection via a  
“srtpreplayprotection” option in rtp.conf. The default is   
  
“yes”   
  

   Affected Versions
Product   Release Series  
  Asterisk Open Source 13.x   13.38.1 
  Asterisk Open Source 16.x   16.16.0 
  Asterisk Open Source 17.x   17.9.1  
  Asterisk Open Source 18.x   18.2.0  
   Certified Asterisk  16.x   16.8-cert5  

  Corrected In
   Product  Release   
Asterisk Open Source   13.38.2, 16.16.1, 17.9.2, 18.2.1   
 Certified Asterisk   16.8-cert6  

 Patches 
   Patch URL  Revision  
   https:/downloads.asterisk.org/pub/security/AST-2021-003-13.diff   13.38.2
   https:/downloads.asterisk.org/pub/security/AST-2021-003-16.diff   16.16.1
   https:/downloads.asterisk.org/pub/security/AST-2021-003-17.diff   17.9.2 
   https:/downloads.asterisk.org/pub/security/AST-2021-003-18.diff   18.2.1 
   https:/downloads.asterisk.org/pub/security/AST-2021-003-16.8.diff Certified  
 Asterisk   
 16.8-cert6 

 Links   https://issues.asterisk.org/jira/browse/ASTERISK-29260   
  
 https://downloads.asterisk.org/pub/security/AST-2021-003.html

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security  
  
This document may be superseded by later versions; if so, the latest  
version will be posted at 
https://downloads.digium.com/pub/security/AST-2021-003.pdf and
https://downloads.digium.com/pub/security/AST-2021-003.html   

Revision History
  Date  Editor Revisions Made 
February 4, 2021   George Joseph Initial  
February 5, 2021   George Joseph Added CVE ID 

   Asterisk Project Security Advisory - AST-2021-003
   Copyright © 2021 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
   original, unaltered form.

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[asterisk-users] AST-2021-002: Remote crash possible when negotiating T.38

2021-02-18 Thread Asterisk Security Team
   Asterisk Project Security Advisory - AST-2021-002

  Product Asterisk
  Summary Remote crash possible when negotiating T.38 
 Nature of Advisory   Denial of service   
   Susceptibility Remote authenticated sessions   
  SeverityMinor   
   Exploits Known No  
Reported On   December 8, 2020
Reported By   Gregory Massel  
 Posted On
  Last Updated On February 5, 2021
  Advisory Contactkharwell AT sangoma DOT com 
  CVE NameCVE-2021-26717  

  Description When re-negotiating for T.38 if the initial remote  
  response was delayed just enough Asterisk would send
  both audio and T.38 in the SDP. If this happened, and   
  the remote responded with a declined T.38 stream then   
  Asterisk would crash.   
Modules Affected  res_pjsip_session.c, res_pjsip_t38.c

Resolution  When re-negotiating for T.38, and a delay occurs Asterisk 
now sends SDP only for the expected T.38 stream. A check was  
also put in place to ensure an active T.38 media stream is
active within Asterisk when attempting to change state for
fax.  

   Affected Versions
Product   Release Series  Introduced  
  Asterisk Open Source 16.x   16.15.0 
  Asterisk Open Source 17.x   17.9.0  
  Asterisk Open Source 18.x   18.1.0  
   Certified Asterisk  16.8   16.8-cert4  

  Corrected In
 Product  Release 
   Asterisk Open Source   16.16.1, 17.9.2, 18.2.1 
Certified Asterisk   16.8-cert6   

  Patches 
   Patch URL   Revision 
 
   https://downloads.asterisk.org/pub/security/AST-2021-002-16.diff   Asterisk  
 
  16
 
   https://downloads.asterisk.org/pub/security/AST-2021-002-17.diff   Asterisk  
 
  17
 
   https://downloads.asterisk.org/pub/security/AST-2021-002-18.diff   Asterisk  
 
  18
 
   https://downloads.asterisk.org/pub/security/AST-2021-002-16.8.diff Certified 
 
  Asterisk  
 
  
16.8-cert6 

 Links   https://issues.asterisk.org/jira/browse/ASTERISK-29203   
  
 https://downloads.asterisk.org/pub/security/AST-2021-002.html

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security  
  
This document may be superseded by later versions; if so, the latest  
version will be posted at 
http://downloads.digium.com/pub/security/AST-2021-002.pdf and 
http://downloads.digium.com/pub/security/AST-2021-002.html

Revision History
   Date  EditorRevisions Made 
February 1, 2021 Kevin Harwell   Initial revision 

   Asterisk Project Security Advisory - AST-2021-002
   Copyright © 2021 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
   original, unaltered form.

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[asterisk-users] AST-2021-001: Remote crash in res_pjsip_diversion

2021-02-18 Thread Asterisk Security Team
   Asterisk Project Security Advisory - AST-2021-001

  Product Asterisk
  Summary Remote crash in res_pjsip_diversion 
 Nature of Advisory   Denial of service   
   Susceptibility Remote authenticated sessions   
  SeverityModerate
   Exploits Known No  
Reported On   December 28 2020
Reported By   Ivan Poddubny   
 Posted OnJanuary 04 2021 
  Last Updated On January 04 2021 
  Advisory Contactgjoseph AT sangoma DOT com  
  CVE NameCVE-2020-35776  

  Description If a registered user is tricked into dialing a  
  malicious  number that sends lots of 181 responses to   
  Asterisk, each one will cause a 181 to be sent back to  
  the original caller with an increasing number of
  entries in the “Supported” header. Eventually the 
  
  number of entries in the header exceeds the size of 
  the entry array and causes a crash. 
Modules Affected  res_pjsip_diversion.c   

Resolution  Before updating the “Supported” header with a new entry,
  
Asterisk now checks that the entry doesn’t already exist and  
that adding an entry won’t exceed the size of the entry   
array.

   Affected Versions
 Product   Release Series  
  Asterisk Open Source  13.X   13.38.1
  Asterisk Open Source  16.X   16.15.1
  Asterisk Open Source  17.X   17.9.1 
  Asterisk Open Source  18.X   18.1.1 

  Corrected In
   Product  Release   
Asterisk Open Source   13.38.2, 16.16.1, 17.9.2, 18.2.1   

Patches 
  Patch URL Revision  
https://downloads.digium.com/pub/security/AST-2021-001-13.diff  13.38.2   
https://downloads.digium.com/pub/security/AST-2021-001-16.diff  16.16.1   
https://downloads.digium.com/pub/security/AST-2021-001-17.diff  17.9.2
https://downloads.digium.com/pub/security/AST-2021-001-18.diff  18.2.1

 Links   https://issues.asterisk.org/jira/browse/ASTERISK-29227   
 https://downloads.asterisk.org/pub/security/AST-2021-001.html

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security  
  
This document may be superseded by later versions; if so, the latest  
version will be posted at 
https://downloads.digium.com/pub/security/AST-2021-001.pdf and
https://downloads.digium.com/pub/security/AST-2021-001.html   

Revision History
 Date Editor   Revisions Made 
December 29, 2020   George JosephInitial revision 

   Asterisk Project Security Advisory - AST-2021-001
   Copyright © 2020 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
   original, unaltered form.

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[asterisk-users] Asterisk 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6 Now Available (Security)

2021-02-18 Thread Asterisk Development Team
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.

These releases are available for immediate download at

https://downloads.asterisk.org/pub/telephony/asterisk/releases
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases

The following security vulnerabilities were resolved in these versions:

* AST-2021-001: Remote crash in res_pjsip_diversion
  If a registered user is tricked into dialing a

* AST-2021-002: Remote crash possible when negotiating T.38
  When

* AST-2021-003: Remote attacker could prematurely tear down SRTP calls
  An unauthenticated remote attacker could replay SRTP packets which could cause
  an Asterisk instance configured without strict RTP validation to tear down
  calls prematurely.

* AST-2021-004: An unsuspecting user could crash Asterisk with multiple
hold/unhold requests
  Due to a signedness comparison mismatch, an authenticated WebRTC client could
  cause a stack overflow and Asterisk crash by sending multiple hold/unhold
  requests in quick succession.

* AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver
  Given a scenario where an outgoing call is placed from Asterisk to a remote
  SIP server it is possible for a crash to occur.

For a full list of changes in the current releases, please see the ChangeLogs:

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
https://downloads.asterisk.org/pub/security/AST-2021-005.pdf

Thank you for your continued support of Asterisk!-- 
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