[asterisk-users] Asterisk 15 Beta Released

[Matt Fredrickson]

It is with great pleasure I wish to inform you of the first beta
release of the new Asterisk 15 branch. It's a very exciting time to be
a user of Asterisk! Asterisk 15 is arguably the biggest release of
Asterisk that has happened in the last 10 or so years. There has been
a lot of work done in the Asterisk core to better support newer
multi-stream video and WebRTC related technologies.  For those who are
interested, much of this will be covered in blog posts at
http://blogs.asterisk.org/ over the next month or two.

Typically, when a new major branch of Asterisk is created (13, 14,
15...), there are a few months of testing on the new branch that
occurs prior to release in order to find regressions and other issues
that may cause a first official release from the branch to be dead on
arrival for a significant number of users. With today's release of
15.0.0-beta1, this process has begun. Please feel free to start
testing this version of Asterisk in as many adverse environments as
possible. Any bugs should be reported on the Asterisk issue tracker at
https://issues.asterisk.org/

As a side note, due to many of the core changes in the 15 branch that
have been made since Asterisk 14 was released, it has been decided
that Asterisk 15 will not be an LTS release. For those of you who are
not familiar with the differences between LTS versus standard
releases, you can find more information here [1].

Thanks to all the many Asterisk community members for providing so
much help and support to make Asterisk the great open source project
that it is.

P.S. Binary codecs and other modules distributed by Digium are not
immediately available for 15.0.0-beta1, but should be shortly.

Best wishes to all, and happy testing!

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

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

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Re: [asterisk-users] Asterisk 15.0.0-beta1 Now Available

[Joshua Colp]

On Wed, Aug 2, 2017, at 02:28 PM, Ira wrote:
> Re: [asterisk-users] Asterisk 15.0.0-beta1 Now AvailableHello Asterisk,
> 
>  Wednesday, August 2, 2017, 9:20:19 AM, you wrote:
> 
> 
>  > The Asterisk Development Team would like to announce the first beta
>  > of Asterisk 15.0.0. This beta is available for immediate download at
>  > http://downloads.asterisk.org/pub/telephony/asterisk The release of
>  > Asterisk 15.0.0-beta1 resolves several issues reported by the
>  > community and would have not been possible without your
>  > participation.
> 
>  In the interest of helping with the beta, maybe including a link to
>  where to report bugs would be useful? I downloaded it but it fails to
>  compile and ends with this error:
> 
> *   [CC] app_voicemail.c -> app_voicemail.o cc1: error: unrecognized
> command line option "-Wno-format-truncation" make[1]: ***
> [app_voicemail.o] Error 1 make: *** [apps] Error 2
> 
> *A long time ago I knew where to report problems, but I've no idea where
> that location might be any more.
> 
>  32 bit CentOS final version Don't recall if it's 5 or 6 but I know it's
>  out of support as yum update stopped working.

The address to report issues is https://issues.asterisk.org/jira

-- 
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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] Asterisk 15.0.0-beta1 Now Available

[Ira]

Title: Re: [asterisk-users] Asterisk 15.0.0-beta1 Now Available


Hello Asterisk,

Wednesday, August 2, 2017, 9:20:19 AM, you wrote:


> The Asterisk Development Team would like to announce the first beta of Asterisk 15.0.0.
> This beta is available for immediate download at 
> http://downloads.asterisk.org/pub/telephony/asterisk 
> The release of Asterisk 15.0.0-beta1 resolves several issues reported by the
> community and would have not been possible without your participation.

In the interest of helping with the beta, maybe including a link to where to report bugs would be useful? I downloaded it but it fails to compile and ends with this error:

   [CC] app_voicemail.c -> app_voicemail.o
cc1: error: unrecognized command line option "-Wno-format-truncation"
make[1]: *** [app_voicemail.o] Error 1
make: *** [apps] Error 2

A long time ago I knew where to report problems, but I've no idea where that location might be any more.

32 bit CentOS final version Don't recall if it's 5 or 6 but I know it's out of support as yum update stopped working.

-- Ira


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Re: [asterisk-users] SIP-Header Pass-Through

[John Kiniston]

My only suggestion would be you could reduce your line count by replacing
your GotoIf statements with ExecIF statements.

exten => addheader,1,ExecIf($["x${ARG1}" !=
"x"]?Set(PJSIP_HEADER(add,Route)=${ARG1}))
 same => n,ExecIf($["x${ARG2}" !=
"x"]?Set(PJSIP_HEADER(add,P-Charging-Vector)=${ARG2}))
 same => n,ExecIf($["x${ARG3}" !=
"x"]?Set(PJSIP_HEADER(add,P-Visited-Network-ID)=${ARG3}))
 same => n,ExecIf($["x${ARG4}" !=
"x"]?Set(PJSIP_HEADER(add,X-C-Params)=${ARG4}))
 same => n,ExecIf($["x${ARG5}" != "x"]?Set(PJSIP_HEADER(add,X-URI)=${ARG5}))
 same => n,Return()


On Wed, Aug 2, 2017 at 6:39 AM, Carsten Bock  wrote:

> Hi,
>
> quick question:
> I need to pass-through some headers from the A-Leg to the B-Leg, which
> are connected using PJSIP.
> Currently I do the following:
>
> [handler]
> exten => addheader,1,GotoIf($["${ARG1}" == ""]?3)
> exten => addheader,2,Set(PJSIP_HEADER(add,Route)=${ARG1})
> exten => addheader,3,GotoIf($["${ARG2}" == ""]?5)
> exten => addheader,4,Set(PJSIP_HEADER(add,P-Charging-Vector)=${ARG2})
> exten => addheader,5,GotoIf($["${ARG3}" == ""]?7)
> exten => addheader,6,Set(PJSIP_HEADER(add,P-Visited-Network-ID)=${ARG3})
> exten => addheader,7,GotoIf($["${ARG4}" == ""]?9)
> exten => addheader,8,Set(PJSIP_HEADER(add,X-C-Params)=${ARG4})
> exten => addheader,9,GotoIf($["${ARG5}" == ""]?11)
> exten => addheader,10,Set(PJSIP_HEADER(add,X-URI)=${ARG5})
> exten => addheader,11,Return
>
> [pcscf1]
> exten => _.,1,Dial(PJSIP/pcscf1/sip:${EXTEN}@${SIPDOMAIN},,b(
> handler^addheader^1,(${PJSIP_HEADER(read,Route)},${PJSIP_
> HEADER(read,P-Charging-Vector)},${PJSIP_HEADER(read,P-
> Visited-Network-ID)},${PJSIP_HEADER(read,X-C-Params)},${
> PJSIP_HEADER(read,X-URI)})))
>
> This works, but you'll have to admit, it's not the most
> readable-Version of an extensions.conf.
>
> Are there any better solutions around?
>
> Thanks,
> Carsten
>
> P.S.: I may have found an issue in Asterisk 13.x (LTS); if I send the
> right SIP-Message I can make it crash. I'm currently verifying, if it
> still happens with the latest 13.x release from GIT. If I can confirm
> it with the latest 13.x from GIT, should I post my findings to the
> dev-list or is there some security mailing list, as this would apply
> to all installations around there?
>
> --
> Carsten Bock
> CEO (Geschäftsführer)
>
> ng-voice GmbH
> Millerntorplatz 1
> 20359 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:cars...@ng-voice.com
>
> Office +49 40 5247593-40
> Fax +49 40 5247593-99
>
> Sitz der Gesellschaft: Hamburg
> Registergericht: Amtsgericht Hamburg, HRB 120189
> Geschäftsführer: Carsten Bock
> Ust-ID: DE279344284
>
> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
> http://www.ng-voice.com/imprint/
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> 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




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build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
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Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

[Carlos Chavez]

On 2017-08-02 07:08, Nathan Anderson wrote:

Richard Kenner wrote:


But the question here
was *Asterisk*, not kernels.  User-level code has *way* fewer
dependencies.


*Precisely*.  Unless we're talking DAHDI here (which we're not), Linux
& ESXi are red herrings.

Carlos Chavez wrote:


  I am having a very tough time trying to replace an Elastix 2.X
install running as a virtual machine on ESXI 4.


There's no way this has anything to do with ESXi or the version of it
that you are running.  Zero.  Zip.  Zilch.

If you want to prove this to yourself and others, take the *exact*
same binary bits, install them bare-metal on another piece of
hardware, run the same traffic through it, and watch it crash and burn
in the same way.  The only way that I can see this playing out
differently is if the bug (yes, bug) in Asterisk and associated
libraries is extremely timing-dependent, and running it in a VM is
exposing this bug in a way that most bare-metal installations
wouldn't.


I will try using chan_sip
instead of PJSIP to get things running but confidence is not high.


Given that the log entry you pasted into your e-mail references
"libasteriskpj.so", I'd bet $$$ that switching to chan_sip has an
extremely high likelihood of working, assuming that your set-up has no
particular dependencies on PJSIP-specific features that you have to
work around (and if you are migrating from an Asterisk 1.6
installation, I'm guessing it doesn't).

Best of luck,

-- Nathan


I run the CentOS 7.3 / Asterisk 13.17.0 combination of software 
installed from the same sources on multiple servers across a wide 
variety of hardware (both metal and virtual) and this is the only place 
that I have encountered this particular problem.  That is why the only 
variable left is the version of ESXI as newer versions work.  
Unfortunately I do not have a newer server where I can just import this 
same VM to completely eliminate the possibility.


--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
dCAP #1349
+52 (55)8116-9161

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Re: [asterisk-users] CallerID matching failure, possibly bug.

[Joshua Colp]

On Wed, Aug 2, 2017, at 08:37 AM, Blank Field wrote:
> Hello everyone.
> 
> Seems like i've managed to isolate a troubling behaviour on my asterisk.
> CallerID pattern matching does not work on the first try.
> 
> Technical info below:
> asterisk*CLI> core show version
> Asterisk 14.5.0 built by admin @ asterisk.domain on a x86_64 running
> Linux on 2017-06-13 14:26:54 UTC
> 
> I have an endpoint 616 with CALLERID(name) set to '616' and
> CALLERID(num) set to '616' at the user device. The endpoint is
> registered at asterisk as 616. Contact is 616@endpoint_ip.



This is likely a result of the PJSIP channel driver and not any core
things. It may not be applying the caller ID early enough. Since you
have a case that reproduces it go ahead and file an issue[1].

Cheers,

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

-- 
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] SIP-Header Pass-Through

[Joshua Colp]

On Wed, Aug 2, 2017, at 10:39 AM, Carsten Bock wrote:
> Hi,
> 
> quick question:
> I need to pass-through some headers from the A-Leg to the B-Leg, which
> are connected using PJSIP.
> Currently I do the following:
> 
> [handler]
> exten => addheader,1,GotoIf($["${ARG1}" == ""]?3)
> exten => addheader,2,Set(PJSIP_HEADER(add,Route)=${ARG1})
> exten => addheader,3,GotoIf($["${ARG2}" == ""]?5)
> exten => addheader,4,Set(PJSIP_HEADER(add,P-Charging-Vector)=${ARG2})
> exten => addheader,5,GotoIf($["${ARG3}" == ""]?7)
> exten => addheader,6,Set(PJSIP_HEADER(add,P-Visited-Network-ID)=${ARG3})
> exten => addheader,7,GotoIf($["${ARG4}" == ""]?9)
> exten => addheader,8,Set(PJSIP_HEADER(add,X-C-Params)=${ARG4})
> exten => addheader,9,GotoIf($["${ARG5}" == ""]?11)
> exten => addheader,10,Set(PJSIP_HEADER(add,X-URI)=${ARG5})
> exten => addheader,11,Return
> 
> [pcscf1]
> exten =>
> _.,1,Dial(PJSIP/pcscf1/sip:${EXTEN}@${SIPDOMAIN},,b(handler^addheader^1,(${PJSIP_HEADER(read,Route)},${PJSIP_HEADER(read,P-Charging-Vector)},${PJSIP_HEADER(read,P-Visited-Network-ID)},${PJSIP_HEADER(read,X-C-Params)},${PJSIP_HEADER(read,X-URI)})))
> 
> This works, but you'll have to admit, it's not the most
> readable-Version of an extensions.conf.
> 
> Are there any better solutions around?

Not really.

> 
> Thanks,
> Carsten
> 
> P.S.: I may have found an issue in Asterisk 13.x (LTS); if I send the
> right SIP-Message I can make it crash. I'm currently verifying, if it
> still happens with the latest 13.x release from GIT. If I can confirm
> it with the latest 13.x from GIT, should I post my findings to the
> dev-list or is there some security mailing list, as this would apply
> to all installations around there?

The process for reporting security vulnerabilities is documented on the
wiki[1].

[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Vulnerabilities

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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[asterisk-users] SIP-Header Pass-Through

[Carsten Bock]

Hi,

quick question:
I need to pass-through some headers from the A-Leg to the B-Leg, which
are connected using PJSIP.
Currently I do the following:

[handler]
exten => addheader,1,GotoIf($["${ARG1}" == ""]?3)
exten => addheader,2,Set(PJSIP_HEADER(add,Route)=${ARG1})
exten => addheader,3,GotoIf($["${ARG2}" == ""]?5)
exten => addheader,4,Set(PJSIP_HEADER(add,P-Charging-Vector)=${ARG2})
exten => addheader,5,GotoIf($["${ARG3}" == ""]?7)
exten => addheader,6,Set(PJSIP_HEADER(add,P-Visited-Network-ID)=${ARG3})
exten => addheader,7,GotoIf($["${ARG4}" == ""]?9)
exten => addheader,8,Set(PJSIP_HEADER(add,X-C-Params)=${ARG4})
exten => addheader,9,GotoIf($["${ARG5}" == ""]?11)
exten => addheader,10,Set(PJSIP_HEADER(add,X-URI)=${ARG5})
exten => addheader,11,Return

[pcscf1]
exten => 
_.,1,Dial(PJSIP/pcscf1/sip:${EXTEN}@${SIPDOMAIN},,b(handler^addheader^1,(${PJSIP_HEADER(read,Route)},${PJSIP_HEADER(read,P-Charging-Vector)},${PJSIP_HEADER(read,P-Visited-Network-ID)},${PJSIP_HEADER(read,X-C-Params)},${PJSIP_HEADER(read,X-URI)})))

This works, but you'll have to admit, it's not the most
readable-Version of an extensions.conf.

Are there any better solutions around?

Thanks,
Carsten

P.S.: I may have found an issue in Asterisk 13.x (LTS); if I send the
right SIP-Message I can make it crash. I'm currently verifying, if it
still happens with the latest 13.x release from GIT. If I can confirm
it with the latest 13.x from GIT, should I post my findings to the
dev-list or is there some security mailing list, as this would apply
to all installations around there?

-- 
Carsten Bock
CEO (Geschäftsführer)

ng-voice GmbH
Millerntorplatz 1
20359 Hamburg / Germany

http://www.ng-voice.com
mailto:cars...@ng-voice.com

Office +49 40 5247593-40
Fax +49 40 5247593-99

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
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Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

[Nathan Anderson]

Richard Kenner wrote:

> But the question here
> was *Asterisk*, not kernels.  User-level code has *way* fewer
> dependencies.

*Precisely*.  Unless we're talking DAHDI here (which we're not), Linux & ESXi 
are red herrings.

Carlos Chavez wrote:

>   I am having a very tough time trying to replace an Elastix 2.X 
> install running as a virtual machine on ESXI 4.  

There's no way this has anything to do with ESXi or the version of it that you 
are running.  Zero.  Zip.  Zilch.

If you want to prove this to yourself and others, take the *exact* same binary 
bits, install them bare-metal on another piece of hardware, run the same 
traffic through it, and watch it crash and burn in the same way.  The only way 
that I can see this playing out differently is if the bug (yes, bug) in 
Asterisk and associated libraries is extremely timing-dependent, and running it 
in a VM is exposing this bug in a way that most bare-metal installations 
wouldn't.

> I will try using chan_sip 
> instead of PJSIP to get things running but confidence is not high.

Given that the log entry you pasted into your e-mail references 
"libasteriskpj.so", I'd bet $$$ that switching to chan_sip has an extremely 
high likelihood of working, assuming that your set-up has no particular 
dependencies on PJSIP-specific features that you have to work around (and if 
you are migrating from an Asterisk 1.6 installation, I'm guessing it doesn't).

Best of luck,

-- Nathan

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[asterisk-users] CallerID matching failure, possibly bug.

[Blank Field]

Hello everyone.

Seems like i've managed to isolate a troubling behaviour on my asterisk.
CallerID pattern matching does not work on the first try.

Technical info below:
asterisk*CLI> core show version
Asterisk 14.5.0 built by admin @ asterisk.domain on a x86_64 running
Linux on 2017-06-13 14:26:54 UTC

I have an endpoint 616 with CALLERID(name) set to '616' and
CALLERID(num) set to '616' at the user device. The endpoint is
registered at asterisk as 616. Contact is 616@endpoint_ip.

[cidmatch]
exten => _.,1,NoOp()
exten => _./_6XX,1,SayDigits(1)
same => 2,SayDigits(2)
same => 3,SayDigits(3)
same => 4,SayDigits(4)

asterisk*CLI> dialplan show cidmatch
[ Context 'cidmatch' created by 'pbx_config' ]
  '_.' (CID match '_6XX') =>  1. SayDigits(1)
 [pbx_config]
2. SayDigits(2)
 [pbx_config]
3. SayDigits(3)
 [pbx_config]
4. SayDigits(4)
 [pbx_config]
  '_.' =>   1. NoOp()
 [pbx_config]

-= 2 extensions (5 priorities) in 1 context. =-

Please note two pattern matching attempts.

That way, SayDigits app works, and the digits are played.

If I comment out the first like, matching _., the following situation
happens:

[cidmatch]
;exten => _.,1,NoOp()
exten => _./_6XX,1,SayDigits(1)
same => 2,SayDigits(2)
same => 3,SayDigits(3)
same => 4,SayDigits(4)

asterisk*CLI> dialplan show cidmatch
[ Context 'cidmatch' created by 'pbx_config' ]
  '_.' (CID match '_6XX') =>  1. SayDigits(1)
 [pbx_config]
2. SayDigits(2)
 [pbx_config]
3. SayDigits(3)
 [pbx_config]
4. SayDigits(4)
 [pbx_config]

-= 1 extension (4 priorities) in 1 context. =-

[2017-07-25 15:03:32.037] NOTICE[13524]: res_pjsip_session.c:2141
new_invite: Call from '616' (UDP:IP:PORT) to extension '1' rejected
because extension not found in context 'cidmatch'.

To verify that CALLERID is correct:

[cidmatch]
exten => _.,1,Verbose(1,name: ${CALLERID(name)} num: ${CALLERID(num)})
;exten => _./_6XX,1,SayDigits(1)
same => 2,SayDigits(2)
same => 3,SayDigits(3)
same => 4,SayDigits(4)

asterisk*CLI> dialplan show cidmatch
[ Context 'cidmatch' created by 'pbx_config' ]
  '_.' =>   1. Verbose(1,name: ${CALLERID(name)} num:
${CALLERID(num)}) [pbx_config]
2. SayDigits(2)
 [pbx_config]
3. SayDigits(3)
 [pbx_config]
4. SayDigits(4)
 [pbx_config]

-= 1 extension (4 priorities) in 1 context. =-

 -- Executing [1@cidmatch:1] Verbose("PJSIP/616-0b2f", "1,name:
616 num: 616") in new stack
 name: 616 num: 616
-- Executing [1@cidmatch:2] SayDigits("PJSIP/616-0b2f", "2")
in new stack

That seriously took me some time to investigate.
https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching
The part on "Matching on Caller ID", aka "ex-girlfriend logic" seems
to be broken on my build.

Any advice appreciated, but it works for now.

Regards, Duelist.
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Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

[George Joseph]

On Tue, Aug 1, 2017 at 12:53 PM, Carlos Chavez  wrote:

>  I am having a very tough time trying to replace an Elastix 2.X
> install running as a virtual machine on ESXI 4.  I tried using the Freepbx
> 14 ISO that installs CentOS 6 along with Asterisk 13.16 but I keep getting
> random segfaults:
>
> [175711.476685] asterisk[2942]: segfault at 188 ip 7fc6c41abffc sp
> 7fc608575890 error 4 in libasteriskpj.so.2[7fc6c4144000+14c000]
>

The messages that get dumped to the kernel log aren't of much use.  See
below for more info.


>
>  I then proceeded to install a CentOS 7.3 VM and compiled Asterisk
> 13.17.0 by hand.


That *should* be a good combination.


> We are still using Freepbx 14 for the front end.  We did some testing over
> the weekend and calls were coming in and out and all extensions were
> registered.  Come Monday Asterisk started segfaulting again with exactly
> the same error.  Maybe VMware is too old to support the newer CentOS and
> Asterisk?  The Elastix install is based on CentOS 5 and Asterisk 1.6.  I
> have no idea how to approach this.  It only segfaults when there are only
> more than a couple simultaneous calls, that is why testing with only a
> couple of calls worked.
>
>  There are several core dump files but I really do not know how to use
> them for debugging Asterisk.  Any ideas?


If you still have the coredump files, you can use the ast_coredumper
utility located in /var/lib/asterisk/scripts to extract the human-readable
stack traces.  "sudo /var/lib/asterisk/scripts/ast_coredumper --help" will
get you more info on how to run the command.  The *-thread1.txt file it
produces will be the most helpful but all 4 should be attached to the
Asterisk issue should you decide to create one.



> I will try using chan_sip instead of PJSIP to get things running but
> confidence is not high.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> dCAP #1349
> +52 (55)8116-9161
>
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> Check out the new Asterisk community forum at:
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>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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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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Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

[Richard Kenner]

> There are certain versions of the Linux kernel that have no support
> under the older version of ESXI.  We started having issues under our
> ESXI v4 setup with RH Enterprise and vmware's response was, "It's
> not supported"

"not supported" and "does not work" are not the same thing.  ESXI
emulates specific hardware.  Most kernels will work with old hardware,
so they should work with old ESXI, though there may need to be some
configuration changes and there's always the possibility of bugs in
ESXI that weren't detected by older kernels.  But the question here
was *Asterisk*, not kernels.  User-level code has *way* fewer
dependencies.

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Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

[Doug Lytle]

>>> On Aug 2, 2017, at 6:45 AM, Richard Kenner ken...@gnat.com wrote:

>>> I wouldn't believe it either.  I'd be quite surprised if something won't
>>> work with any ESXI version.  *Perhaps* there's a configuration issue, but
>>> I'd be surprised about that too.

There are certain versions of the Linux kernel that have no support under the 
older version of ESXI.  We started having issues under our ESXI v4 setup with 
RH Enterprise and vmware's response was, "It's not supported"

Doug

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Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

[Richard Kenner]

> The version is licensed and the customer does not want to invest on new 
> hardware/software at the moment.  If the ESXI version is too old I need 
> to give them definitive proof that the segfaults are caused by that but 
> since the old elastix has been running there for years they do not quite 
> believe it.

I wouldn't believe it either.  I'd be quite surprised if something won't
work with any ESXI version.  *Perhaps* there's a configuration issue, but
I'd be surprised about that too.

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