astlinux-users — AstLinux Users Mailing List

Re: [Astlinux-users] Getting high processor load on ALIX box

[Michael K. <mic...@ip...>]

So do you think I should turn it off by default and only turn it on when I need it?

Regards
Michael Knill

From: Michael Knill <mic...@ip...>
Reply-To: AstLinux List <ast...@li...>
Date: Wednesday, 30 August 2017 at 9:01 am
To: AstLinux List <ast...@li...>
Subject: [Astlinux-users] Getting high processor load on ALIX box

Hmm I liked it better before I implemented Monit. Blissful ignorance ☺

I got am email from one of my servers with the following:
Event:                    Resource limit matched
                                                                Service    3016-Tilton-CM1
                Date:        Tue, 29 Aug 2017 16:26:14
                Action:      alert
                Host:        3016-Tilton-CM1
                Description: loadavg(5min) of 2.2 matches resource limit [loadavg(5min) > 2.0]

I found this in the logs of the server:
Aug 29 16:04:23 3016-Tilton-CM1 user.info kernel: AIF:Port 0 OS fingerprint: IN=ppp0 OUT= MAC= SRC=221.122.59.98 DST=115.187.184.60 LEN=69 TOS=0x00 PREC=0x00 TTL=44 ID=31131 PROTO=UDP SPT=6973 DPT=0 LEN=49
Aug 29 16:04:56 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: pcap_dispatch took too long (took 1023275970 nsec, over threshold of 1000000000 nsec)
Aug 29 16:04:56 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: event processing took longer than a second (took 1319566069 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:02 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: pcap_dispatch took too long (took 1274262839 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:02 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: event processing took longer than a second (took 1461235038 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:04 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: event processing took longer than a second (took 1057674551 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:11 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: event processing took longer than a second (took 1513263031 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:14 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: pcap_dispatch took too long (took 1548211950 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:14 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: event processing took longer than a second (took 2129665587 nsec, over threshold of 1000000000 nsec)

..... and lots more

Should I turn off the Netstat server? I assume this turns off darkstat!

Regards
Michael Knill

Re: [Astlinux-users] Can't mount CDROM or USB mem stick

[Tim T. <tt...@z-...>]

Thanks, Bitvise worked.

 

Just out of curiosity, shouldn’t I be able to perform ftp to the AstLinux server?

Also, what would be the proper way to mount a SATA CDROM drive?

 

From: John Novack [mailto:jn...@co...] 
Sent: Tuesday, August 29, 2017 8:23 PM
To: AstLinux Users Mailing List
Subject: Re: [Astlinux-users] Can't mount CDROM or USB mem stick

 

Think out of the ( linux ) box.
One easy solution is to use Bitvise ( https://www.bitvise.com ) a free solution to use from a windows machine
Presents with a dual file transfer panel, and an SSH command window.
Requires no changes to the AstLinux system, though I always change the SSH port if there is off LAN access.
I have used it for some time to support a bunch of remote thin clients on the collectors peer to peer network.

Be sure to check permissions on files transferred, as they may not be what you want.

I feel sure there are other solutions as well.

John Novack 

Tim Turpin wrote:

Sorry to be a pest, still a noob here.  I need to be able to copy some .wav files from a Windows system to AstLinux.  I tried FTP, but I can’t find it on the AstLinux system.  I was going to download it, but couldn’t find YUM and wget kept failing.  I then tried to read from a FAT formatted USB stick, but couldn’t get it to mount.  Then I tried a USB CDROM, same thing.  I now have a SATA CD ROM plugged in, and it doesn’t show up in fdisk –l.

Am I wrong in assuming that Linux commands should work as they do in Centos?






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https://lists.sourceforge.net/lists/listinfo/astlinux-users
 
Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....





-- 
 
Dog is my Co-pilot

Re: [Astlinux-users] Can't mount CDROM or USB mem stick

[John N. <jn...@co...>]

Think out of the ( linux ) box.
One easy solution is to use Bitvise ( https://www.bitvise.com ) a free solution to use from a windows machine
Presents with a dual file transfer panel, and an SSH command window.
Requires no changes to the AstLinux system, though I always change the SSH port if there is off LAN access.
I have used it for some time to support a bunch of remote thin clients on the collectors peer to peer network.

Be sure to check permissions on files transferred, as they may not be what you want.

I feel sure there are other solutions as well.

John Novack

Tim Turpin wrote:
>
> Sorry to be a pest, still a noob here.  I need to be able to copy some .wav files from a Windows system to AstLinux.  I tried FTP, but I can’t find it on the AstLinux system.  I was going to download it, but couldn’t find YUM and wget kept failing.  I then tried to read from a FAT formatted USB stick, but couldn’t get it to mount.  Then I tried a USB CDROM, same thing.  I now have a SATA CD ROM plugged in, and it doesn’t show up in fdisk –l.
>
> Am I wrong in assuming that Linux commands should work as they do in Centos?
>
>
>
> ------------------------------------------------------------------------------
> Check out the vibrant tech community on one of the world's most
> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
>
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....

-- 

Dog is my Co-pilot

[Astlinux-users] Can't mount CDROM or USB mem stick

[Tim T. <tt...@z-...>]

Sorry to be a pest, still a noob here.  I need to be able to copy some .wav
files from a Windows system to AstLinux.  I tried FTP, but I can't find it
on the AstLinux system.  I was going to download it, but couldn't find YUM
and wget kept failing.  I then tried to read from a FAT formatted USB stick,
but couldn't get it to mount.  Then I tried a USB CDROM, same thing.  I now
have a SATA CD ROM plugged in, and it doesn't show up in fdisk -l.

Am I wrong in assuming that Linux commands should work as they do in Centos?

[Astlinux-users] Getting high processor load on ALIX box

[Michael K. <mic...@ip...>]

Hmm I liked it better before I implemented Monit. Blissful ignorance ☺

I got am email from one of my servers with the following:
Event:                    Resource limit matched
                                                                Service    3016-Tilton-CM1
                Date:        Tue, 29 Aug 2017 16:26:14
                Action:      alert
                Host:        3016-Tilton-CM1
                Description: loadavg(5min) of 2.2 matches resource limit [loadavg(5min) > 2.0]

I found this in the logs of the server:
Aug 29 16:04:23 3016-Tilton-CM1 user.info kernel: AIF:Port 0 OS fingerprint: IN=ppp0 OUT= MAC= SRC=221.122.59.98 DST=115.187.184.60 LEN=69 TOS=0x00 PREC=0x00 TTL=44 ID=31131 PROTO=UDP SPT=6973 DPT=0 LEN=49
Aug 29 16:04:56 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: pcap_dispatch took too long (took 1023275970 nsec, over threshold of 1000000000 nsec)
Aug 29 16:04:56 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: event processing took longer than a second (took 1319566069 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:02 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: pcap_dispatch took too long (took 1274262839 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:02 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: event processing took longer than a second (took 1461235038 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:04 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: event processing took longer than a second (took 1057674551 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:11 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: event processing took longer than a second (took 1513263031 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:14 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: pcap_dispatch took too long (took 1548211950 nsec, over threshold of 1000000000 nsec)
Aug 29 16:05:14 3016-Tilton-CM1 daemon.debug darkstat[1424]: WARNING: event processing took longer than a second (took 2129665587 nsec, over threshold of 1000000000 nsec)

..... and lots more

Should I turn off the Netstat server? I assume this turns off darkstat!

Regards
Michael Knill

Re: [Astlinux-users] ASTLinux stopped booting

[Lonnie A. <li...@lo...>]

Hi David,

Agreed, no e2fsck output when it is forced to work for a long period is not ideal.  Thankfully this is a very rare occurrence.

If we can come up with a clever spinning wheel or such counting the lines e2fsck generates might be a good addition.

Recall this all occurs in the "initrd", so the available packages and BusyBox config is quite limited.

Any further discussion on this idea should move to the astlinux-devel list.

Lonnie


On Aug 29, 2017, at 7:48 AM, David Kerr <Da...@Ke...> wrote:

> Is that a good idea (redirecting e2fsck to null)?  15 minutes is a long time to wait with no indication of anything happening.
> 
> David
> 
> On Tue, Aug 29, 2017 at 8:33 AM, Lonnie Abelbeck <li...@lo...> wrote:
> Hi Tim,
> 
> Yes, it was running "e2fsck" with stdout redirected to /dev/null so you did not see it working ... eventually e2fsck will return with a result code if left long enough.
> 
> Good to hear your filesystem is now clean, you might reboot once again to make sure the filesystem is good.
> 
> BTW, if the automatic e2fsck repair did not work, here is the manual fallback procedure:
> --
> ## reboot, and quickly when the RUNNIX boot menu appears type "shell"
> boot: shell
> 
> ## Wait for a "runnix# " CLI prompt.
> ## determine the first Linux ext2 partition, usually always /dev/sda2
> runnix# findfs LABEL=ASTURW
> 
> ## using the findfs result, run e2fsck manually, you may want to add -y or -p options
> runnix# e2fsck /dev/sda2
> 
> ## output a list of options for e2fsck
> runnix# e2fsck
> Usage: e2fsck [-panyrcdfktvDFV] [-b superblock] [-B blocksize]
>                 [-l|-L bad_blocks_file] [-C fd] [-j external_journal]
>                 [-E extended-options] [-z undo_file] device
> 
> Emergency help:
>  -p                   Automatic repair (no questions)
>  -n                   Make no changes to the filesystem
>  -y                   Assume "yes" to all questions
>  -c                   Check for bad blocks and add them to the badblock list
>  -f                   Force checking even if filesystem is marked clean
>  -v                   Be verbose
>  -b superblock        Use alternative superblock
>  -B blocksize         Force blocksize when looking for superblock
>  -j external_journal  Set location of the external journal
>  -l bad_blocks_file   Add to badblocks list
>  -L bad_blocks_file   Set badblocks list
>  -z undo_file         Create an undo file
> 
> ## Depending on how you initially configured AstLinux, check if you also have a ASTKD partition
> ## (typically /dev/sda3 if it exists)
> runnix# findfs LABEL=ASTKD
> ## If the ASTKD label exists repeat the e2fsck steps above using the ASTKD label partition.
> 
> ## reboot by issuing "exit"
> runnix# exit
> --
> 
> Lonnie
> 
> 
> On Aug 29, 2017, at 7:02 AM, Tim Turpin <tt...@z-...> wrote:
> 
> > Never mind.  After sitting at that point for about 15 minutes, it somehow recovered and finished booting up.  All seems well now.
> >
> > From: Tim Turpin [mailto:tt...@z-...]
> > Sent: Tuesday, August 29, 2017 7:34 AM
> > To: ast...@li...
> > Subject: [Astlinux-users] ASTLinux stopped booting
> >
> > We took a power hit on our test system yesterday, and now it will only load up to a certain point and stops with the following screen:
> >
> > <image001.png>
> >
> 

[Astlinux-users] VirtualBox with AstLinux guest

[Lonnie A. <li...@lo...>]

Hi Cody,

I just tried the latest VirtualBox on my Mac OS X system and I was able to use the latest "Guest VM x86-64bit (Video Console):" Install ISO and our instructions, and it all worked as expected.

Your "error zeroing first 1 GB" would imply the VirtualBox virtual drive was not created properly ... Hmmmm not sure how you could have gone wrong there.

I like to keep the install ISO and the created VirtualBox files in the same folder.

Give it another try.

Lonnie


On Aug 28, 2017, at 10:02 PM, Cody Alderson <ald...@gm...> wrote:

> Lonnie,
> 
> This may require a new post, but I get an "error zeroing first 1 GB" when installing Astlinux on Virtual Box. Do you have any idea what may cause that? The install fails because of it.
> 
> -Cody
> 
> On Mon, Aug 28, 2017 at 12:50 PM, Lonnie Abelbeck <li...@lo...> wrote:
> Hi Cody, (comments inline)
> 
> On Aug 28, 2017, at 10:44 AM, Cody Alderson <ald...@gm...> wrote:
> 
> > Can you recommend a book for a Windows guy that goes back to the days of DOS to understand Linux better?
> 
> Googling I found this page: (a part of a college class)
> http://www.cis.rit.edu/class/simg211/unixintro/Filesystem.html
> 
> Though if you want to learn Linux by doing it would be best for you to install
> VirtualBox [ https://www.virtualbox.org/wiki/Downloads ] on Windows and then install
> AstLinux [ https://doc.astlinux-project.org/userdoc:guest_vm_virtualbox ] as a guest of your Windows VirtualBox host.
> 
> Playing with AstLinux as a guest of the VirtualBox VM so you can't do any damage to your production AstLinux system.  If you mess-up you can easily reinstall from the ISO and try again.
> 

Re: [Astlinux-users] ASTLinux stopped booting

[David K. <da...@ke...>]

Is that a good idea (redirecting e2fsck to null)?  15 minutes is a long
time to wait with no indication of anything happening.

David

On Tue, Aug 29, 2017 at 8:33 AM, Lonnie Abelbeck <li...@lo...>
wrote:

> Hi Tim,
>
> Yes, it was running "e2fsck" with stdout redirected to /dev/null so you
> did not see it working ... eventually e2fsck will return with a result code
> if left long enough.
>
> Good to hear your filesystem is now clean, you might reboot once again to
> make sure the filesystem is good.
>
> BTW, if the automatic e2fsck repair did not work, here is the manual
> fallback procedure:
> --
> ## reboot, and quickly when the RUNNIX boot menu appears type "shell"
> boot: shell
>
> ## Wait for a "runnix# " CLI prompt.
> ## determine the first Linux ext2 partition, usually always /dev/sda2
> runnix# findfs LABEL=ASTURW
>
> ## using the findfs result, run e2fsck manually, you may want to add -y or
> -p options
> runnix# e2fsck /dev/sda2
>
> ## output a list of options for e2fsck
> runnix# e2fsck
> Usage: e2fsck [-panyrcdfktvDFV] [-b superblock] [-B blocksize]
>                 [-l|-L bad_blocks_file] [-C fd] [-j external_journal]
>                 [-E extended-options] [-z undo_file] device
>
> Emergency help:
>  -p                   Automatic repair (no questions)
>  -n                   Make no changes to the filesystem
>  -y                   Assume "yes" to all questions
>  -c                   Check for bad blocks and add them to the badblock
> list
>  -f                   Force checking even if filesystem is marked clean
>  -v                   Be verbose
>  -b superblock        Use alternative superblock
>  -B blocksize         Force blocksize when looking for superblock
>  -j external_journal  Set location of the external journal
>  -l bad_blocks_file   Add to badblocks list
>  -L bad_blocks_file   Set badblocks list
>  -z undo_file         Create an undo file
>
> ## Depending on how you initially configured AstLinux, check if you also
> have a ASTKD partition
> ## (typically /dev/sda3 if it exists)
> runnix# findfs LABEL=ASTKD
> ## If the ASTKD label exists repeat the e2fsck steps above using the ASTKD
> label partition.
>
> ## reboot by issuing "exit"
> runnix# exit
> --
>
> Lonnie
>
>
> On Aug 29, 2017, at 7:02 AM, Tim Turpin <tt...@z-...> wrote:
>
> > Never mind.  After sitting at that point for about 15 minutes, it
> somehow recovered and finished booting up.  All seems well now.
> >
> > From: Tim Turpin [mailto:tt...@z-...]
> > Sent: Tuesday, August 29, 2017 7:34 AM
> > To: ast...@li...
> > Subject: [Astlinux-users] ASTLinux stopped booting
> >
> > We took a power hit on our test system yesterday, and now it will only
> load up to a certain point and stops with the following screen:
> >
> > <image001.png>
> >
>
>
>
> ------------------------------------------------------------
> ------------------
> Check out the vibrant tech community on one of the world's most
> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>

Re: [Astlinux-users] ASTLinux stopped booting

[Tim T. <tt...@z-...>]

Thanks for the response.  I still have a lot to learn and you guys have been
great in helping me get this initial system up and running.

-----Original Message-----
From: Lonnie Abelbeck [mailto:li...@lo...] 
Sent: Tuesday, August 29, 2017 8:34 AM
To: AstLinux Users Mailing List
Subject: Re: [Astlinux-users] ASTLinux stopped booting

Hi Tim,

Yes, it was running "e2fsck" with stdout redirected to /dev/null so you did
not see it working ... eventually e2fsck will return with a result code if
left long enough.

Good to hear your filesystem is now clean, you might reboot once again to
make sure the filesystem is good.

BTW, if the automatic e2fsck repair did not work, here is the manual
fallback procedure:
--
## reboot, and quickly when the RUNNIX boot menu appears type "shell"
boot: shell

## Wait for a "runnix# " CLI prompt.
## determine the first Linux ext2 partition, usually always /dev/sda2
runnix# findfs LABEL=ASTURW

## using the findfs result, run e2fsck manually, you may want to add -y or
-p options runnix# e2fsck /dev/sda2

## output a list of options for e2fsck
runnix# e2fsck
Usage: e2fsck [-panyrcdfktvDFV] [-b superblock] [-B blocksize]
		[-l|-L bad_blocks_file] [-C fd] [-j external_journal]
		[-E extended-options] [-z undo_file] device

Emergency help:
 -p                   Automatic repair (no questions)
 -n                   Make no changes to the filesystem
 -y                   Assume "yes" to all questions
 -c                   Check for bad blocks and add them to the badblock list
 -f                   Force checking even if filesystem is marked clean
 -v                   Be verbose
 -b superblock        Use alternative superblock
 -B blocksize         Force blocksize when looking for superblock
 -j external_journal  Set location of the external journal
 -l bad_blocks_file   Add to badblocks list
 -L bad_blocks_file   Set badblocks list
 -z undo_file         Create an undo file

## Depending on how you initially configured AstLinux, check if you also
have a ASTKD partition ## (typically /dev/sda3 if it exists) runnix# findfs
LABEL=ASTKD ## If the ASTKD label exists repeat the e2fsck steps above using
the ASTKD label partition.

## reboot by issuing "exit"
runnix# exit
--

Lonnie


On Aug 29, 2017, at 7:02 AM, Tim Turpin <tt...@z-...> wrote:

> Never mind.  After sitting at that point for about 15 minutes, it somehow
recovered and finished booting up.  All seems well now.
>  
> From: Tim Turpin [mailto:tt...@z-...]
> Sent: Tuesday, August 29, 2017 7:34 AM
> To: ast...@li...
> Subject: [Astlinux-users] ASTLinux stopped booting
>  
> We took a power hit on our test system yesterday, and now it will only
load up to a certain point and stops with the following screen:
>  
> <image001.png>
> 



----------------------------------------------------------------------------
--
Check out the vibrant tech community on one of the world's most engaging
tech sites, Slashdot.org! http://sdm.link/slashdot
_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to
pa...@kr....

Re: [Astlinux-users] ASTLinux stopped booting

[Lonnie A. <li...@lo...>]

Hi Tim,

Yes, it was running "e2fsck" with stdout redirected to /dev/null so you did not see it working ... eventually e2fsck will return with a result code if left long enough.

Good to hear your filesystem is now clean, you might reboot once again to make sure the filesystem is good.

BTW, if the automatic e2fsck repair did not work, here is the manual fallback procedure:
--
## reboot, and quickly when the RUNNIX boot menu appears type "shell"
boot: shell

## Wait for a "runnix# " CLI prompt.
## determine the first Linux ext2 partition, usually always /dev/sda2
runnix# findfs LABEL=ASTURW

## using the findfs result, run e2fsck manually, you may want to add -y or -p options
runnix# e2fsck /dev/sda2

## output a list of options for e2fsck
runnix# e2fsck 
Usage: e2fsck [-panyrcdfktvDFV] [-b superblock] [-B blocksize]
		[-l|-L bad_blocks_file] [-C fd] [-j external_journal]
		[-E extended-options] [-z undo_file] device

Emergency help:
 -p                   Automatic repair (no questions)
 -n                   Make no changes to the filesystem
 -y                   Assume "yes" to all questions
 -c                   Check for bad blocks and add them to the badblock list
 -f                   Force checking even if filesystem is marked clean
 -v                   Be verbose
 -b superblock        Use alternative superblock
 -B blocksize         Force blocksize when looking for superblock
 -j external_journal  Set location of the external journal
 -l bad_blocks_file   Add to badblocks list
 -L bad_blocks_file   Set badblocks list
 -z undo_file         Create an undo file

## Depending on how you initially configured AstLinux, check if you also have a ASTKD partition
## (typically /dev/sda3 if it exists)
runnix# findfs LABEL=ASTKD
## If the ASTKD label exists repeat the e2fsck steps above using the ASTKD label partition.

## reboot by issuing "exit"
runnix# exit
--

Lonnie


On Aug 29, 2017, at 7:02 AM, Tim Turpin <tt...@z-...> wrote:

> Never mind.  After sitting at that point for about 15 minutes, it somehow recovered and finished booting up.  All seems well now.
>  
> From: Tim Turpin [mailto:tt...@z-...] 
> Sent: Tuesday, August 29, 2017 7:34 AM
> To: ast...@li...
> Subject: [Astlinux-users] ASTLinux stopped booting
>  
> We took a power hit on our test system yesterday, and now it will only load up to a certain point and stops with the following screen:
>  
> <image001.png>
> 

Re: [Astlinux-users] ASTLinux stopped booting

[Tim T. <tt...@z-...>]

Never mind.  After sitting at that point for about 15 minutes, it somehow
recovered and finished booting up.  All seems well now.

 

From: Tim Turpin [mailto:tt...@z-...] 
Sent: Tuesday, August 29, 2017 7:34 AM
To: ast...@li...
Subject: [Astlinux-users] ASTLinux stopped booting

 

We took a power hit on our test system yesterday, and now it will only load
up to a certain point and stops with the following screen:

 



 

Is there any way to recover without having to format and rebuild everything
from scratch?

[Astlinux-users] ASTLinux stopped booting

[Tim T. <tt...@z-...>]

We took a power hit on our test system yesterday, and now it will only load
up to a certain point and stops with the following screen:

 



 

Is there any way to recover without having to format and rebuild everything
from scratch?

Re: [Astlinux-users] My Custom TTS Ulaw Sound Files Delete on Reboot of Thin Client Running Astlinux

[Cody A. <ald...@gm...>]

Lonnie,

This may require a new post, but I get an "error zeroing first 1 GB" when
installing Astlinux on Virtual Box. Do you have any idea what may cause
that? The install fails because of it.

-Cody

On Mon, Aug 28, 2017 at 12:50 PM, Lonnie Abelbeck <li...@lo...
> wrote:

> Hi Cody, (comments inline)
>
> On Aug 28, 2017, at 10:44 AM, Cody Alderson <ald...@gm...>
> wrote:
>
> > Lonnie,
> >
> > Symlink is symbolic link in Linux?
>
> Yes.
>
> > Can you recommend a book for a Windows guy that goes back to the days of
> DOS to understand Linux better?
>
> Googling I found this page: (a part of a college class)
> http://www.cis.rit.edu/class/simg211/unixintro/Filesystem.html
>
> Though if you want to learn Linux by doing it would be best for you to
> install
> VirtualBox [ https://www.virtualbox.org/wiki/Downloads ] on Windows and
> then install
> AstLinux [ https://doc.astlinux-project.org/userdoc:guest_vm_virtualbox ]
> as a guest of your Windows VirtualBox host.
>
> Playing with AstLinux as a guest of the VirtualBox VM so you can't do any
> damage to your production AstLinux system.  If you mess-up you can easily
> reinstall from the ISO and try again.
>
>
> > Also, I had previously had to put in the entire path to the ulaw files
> in my dialplan. For example, /var/lib/asterisk/sounds/custom-sounds-filename.
> Now that I have made the custom-sounds persistent, can I just use
> /custom-sounds/filename such as in Playback(/custom-sounds/filename)  ?
>
> You can use Playback(custom-sounds/filename) *without* a leading /
> (slash) as you showed.  Hopefully the class URL above will help you
> understand the UNIX (Linux) file structure.
>
> Also remember that Asterisk's Playback() does not want the sound file
> suffix (ie. no trailing .ulaw or .wav) as Asterisk will look for the best
> matching sound file format for the current channel's CODEC.
>
> In general you should never have to create or delete any files or
> directories outside of the /mnt/kd/ tree path in AstLinux.
>
> Lonnie
>

Re: [Astlinux-users] My Custom TTS Ulaw Sound Files Delete on Reboot of Thin Client Running Astlinux

[Cody A. <ald...@gm...>]

Lonnie,

Thank you for the valuable info!

-Cody

On Mon, Aug 28, 2017 at 12:50 PM, Lonnie Abelbeck <li...@lo...
> wrote:

> Hi Cody, (comments inline)
>
> On Aug 28, 2017, at 10:44 AM, Cody Alderson <ald...@gm...>
> wrote:
>
> > Lonnie,
> >
> > Symlink is symbolic link in Linux?
>
> Yes.
>
> > Can you recommend a book for a Windows guy that goes back to the days of
> DOS to understand Linux better?
>
> Googling I found this page: (a part of a college class)
> http://www.cis.rit.edu/class/simg211/unixintro/Filesystem.html
>
> Though if you want to learn Linux by doing it would be best for you to
> install
> VirtualBox [ https://www.virtualbox.org/wiki/Downloads ] on Windows and
> then install
> AstLinux [ https://doc.astlinux-project.org/userdoc:guest_vm_virtualbox ]
> as a guest of your Windows VirtualBox host.
>
> Playing with AstLinux as a guest of the VirtualBox VM so you can't do any
> damage to your production AstLinux system.  If you mess-up you can easily
> reinstall from the ISO and try again.
>
>
> > Also, I had previously had to put in the entire path to the ulaw files
> in my dialplan. For example, /var/lib/asterisk/sounds/custom-sounds-filename.
> Now that I have made the custom-sounds persistent, can I just use
> /custom-sounds/filename such as in Playback(/custom-sounds/filename)  ?
>
> You can use Playback(custom-sounds/filename) *without* a leading /
> (slash) as you showed.  Hopefully the class URL above will help you
> understand the UNIX (Linux) file structure.
>
> Also remember that Asterisk's Playback() does not want the sound file
> suffix (ie. no trailing .ulaw or .wav) as Asterisk will look for the best
> matching sound file format for the current channel's CODEC.
>
> In general you should never have to create or delete any files or
> directories outside of the /mnt/kd/ tree path in AstLinux.
>
> Lonnie
>
>
> >
> > Thank you so much for your help. I enjoy learning this stuff!
> >
> > -Cody
> >
> > On Mon, Aug 28, 2017 at 9:25 AM, Lonnie Abelbeck <
> li...@lo...> wrote:
> > Hi Cody,
> >
> > Great to hear you got it working.
> >
> > > I don't understand persistent directories yet.
> >
> > As typical with embedded appliances, AstLinux's /var/ mount type is
> "tmpfs" (RAM based, non-persistent).  The flash storage (persistent) is
> mounted type ext2.
> >
> > AstLinux uses a "symbolic link" (ln -s ...) in places to create
> references on the non-persistent /var/ tree that points to persistent
> storage found in the /mnt/kd/ tree as well as to the read-only AstLinux
> ext2 flash image.
> >
> > These "symlinks" can be confusing to follow at first, for example
> following your "custom-sounds" case ...
> > --
> > pbx ~ # ls -l /var/lib/asterisk/sounds/custom-sounds
> > lrwxrwxrwx    1 root     root            38 Aug 27 16:40
> /var/lib/asterisk/sounds/custom-sounds -> /var/tmp/asterisk/sounds/
> custom-sounds
> >
> > pbx ~ # ls -l /var/tmp/asterisk/sounds/custom-sounds
> > lrwxrwxrwx    1 root     root            21 Aug 27 16:41
> /var/tmp/asterisk/sounds/custom-sounds -> /mnt/kd/custom-sounds
> > --
> > Note the " -> " indicates a symlink reference.
> >
> > Remember you should only be adding/editing files in the /mnt/kd/ tree
> path, though if you know a symlink points to the /mnt/kd/ tree path you can
> use the symlink name as a convenience if you wish.
> >
> > Lonnie
>
>
> ------------------------------------------------------------
> ------------------
> Check out the vibrant tech community on one of the world's most
> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>

Re: [Astlinux-users] My Custom TTS Ulaw Sound Files Delete on Reboot of Thin Client Running Astlinux

[Lonnie A. <li...@lo...>]

Hi Cody, (comments inline)

On Aug 28, 2017, at 10:44 AM, Cody Alderson <ald...@gm...> wrote:

> Lonnie,
> 
> Symlink is symbolic link in Linux?

Yes.

> Can you recommend a book for a Windows guy that goes back to the days of DOS to understand Linux better?

Googling I found this page: (a part of a college class)
http://www.cis.rit.edu/class/simg211/unixintro/Filesystem.html

Though if you want to learn Linux by doing it would be best for you to install
VirtualBox [ https://www.virtualbox.org/wiki/Downloads ] on Windows and then install
AstLinux [ https://doc.astlinux-project.org/userdoc:guest_vm_virtualbox ] as a guest of your Windows VirtualBox host.

Playing with AstLinux as a guest of the VirtualBox VM so you can't do any damage to your production AstLinux system.  If you mess-up you can easily reinstall from the ISO and try again.


> Also, I had previously had to put in the entire path to the ulaw files in my dialplan. For example, /var/lib/asterisk/sounds/custom-sounds-filename. Now that I have made the custom-sounds persistent, can I just use /custom-sounds/filename such as in Playback(/custom-sounds/filename)  ?

You can use Playback(custom-sounds/filename) *without* a leading / (slash) as you showed.  Hopefully the class URL above will help you understand the UNIX (Linux) file structure.

Also remember that Asterisk's Playback() does not want the sound file suffix (ie. no trailing .ulaw or .wav) as Asterisk will look for the best matching sound file format for the current channel's CODEC.

In general you should never have to create or delete any files or directories outside of the /mnt/kd/ tree path in AstLinux.

Lonnie


> 
> Thank you so much for your help. I enjoy learning this stuff!
> 
> -Cody
> 
> On Mon, Aug 28, 2017 at 9:25 AM, Lonnie Abelbeck <li...@lo...> wrote:
> Hi Cody,
> 
> Great to hear you got it working.
> 
> > I don't understand persistent directories yet.
> 
> As typical with embedded appliances, AstLinux's /var/ mount type is "tmpfs" (RAM based, non-persistent).  The flash storage (persistent) is mounted type ext2.
> 
> AstLinux uses a "symbolic link" (ln -s ...) in places to create references on the non-persistent /var/ tree that points to persistent storage found in the /mnt/kd/ tree as well as to the read-only AstLinux ext2 flash image.
> 
> These "symlinks" can be confusing to follow at first, for example following your "custom-sounds" case ...
> --
> pbx ~ # ls -l /var/lib/asterisk/sounds/custom-sounds
> lrwxrwxrwx    1 root     root            38 Aug 27 16:40 /var/lib/asterisk/sounds/custom-sounds -> /var/tmp/asterisk/sounds/custom-sounds
> 
> pbx ~ # ls -l /var/tmp/asterisk/sounds/custom-sounds
> lrwxrwxrwx    1 root     root            21 Aug 27 16:41 /var/tmp/asterisk/sounds/custom-sounds -> /mnt/kd/custom-sounds
> --
> Note the " -> " indicates a symlink reference.
> 
> Remember you should only be adding/editing files in the /mnt/kd/ tree path, though if you know a symlink points to the /mnt/kd/ tree path you can use the symlink name as a convenience if you wish.
> 
> Lonnie

Re: [Astlinux-users] My Custom TTS Ulaw Sound Files Delete on Reboot of Thin Client Running Astlinux

[Cody A. <ald...@gm...>]

Lonnie,

Symlink is symbolic link in Linux? Can you recommend a book for a Windows
guy that goes back to the days of DOS to understand Linux better?

Also, I had previously had to put in the entire path to the ulaw files in
my dialplan. For example, /var/lib/asterisk/sounds/custom-sounds-filename.
Now that I have made the custom-sounds persistent, can I just use
/custom-sounds/filename such as in Playback(/custom-sounds/filename)  ?

Thank you so much for your help. I enjoy learning this stuff!

-Cody

On Mon, Aug 28, 2017 at 9:25 AM, Lonnie Abelbeck <li...@lo...>
wrote:

> Hi Cody,
>
> Great to hear you got it working.
>
> > I don't understand persistent directories yet.
>
> As typical with embedded appliances, AstLinux's /var/ mount type is
> "tmpfs" (RAM based, non-persistent).  The flash storage (persistent) is
> mounted type ext2.
>
> AstLinux uses a "symbolic link" (ln -s ...) in places to create references
> on the non-persistent /var/ tree that points to persistent storage found in
> the /mnt/kd/ tree as well as to the read-only AstLinux ext2 flash image.
>
> These "symlinks" can be confusing to follow at first, for example
> following your "custom-sounds" case ...
> --
> pbx ~ # ls -l /var/lib/asterisk/sounds/custom-sounds
> lrwxrwxrwx    1 root     root            38 Aug 27 16:40
> /var/lib/asterisk/sounds/custom-sounds -> /var/tmp/asterisk/sounds/
> custom-sounds
>
> pbx ~ # ls -l /var/tmp/asterisk/sounds/custom-sounds
> lrwxrwxrwx    1 root     root            21 Aug 27 16:41
> /var/tmp/asterisk/sounds/custom-sounds -> /mnt/kd/custom-sounds
> --
> Note the " -> " indicates a symlink reference.
>
> Remember you should only be adding/editing files in the /mnt/kd/ tree
> path, though if you know a symlink points to the /mnt/kd/ tree path you can
> use the symlink name as a convenience if you wish.
>
> Lonnie
>
>
>

Re: [Astlinux-users] My Custom TTS Ulaw Sound Files Delete on Reboot of Thin Client Running Astlinux

[Lonnie A. <li...@lo...>]

Hi Cody,

Great to hear you got it working.

> I don't understand persistent directories yet. 

As typical with embedded appliances, AstLinux's /var/ mount type is "tmpfs" (RAM based, non-persistent).  The flash storage (persistent) is mounted type ext2.

AstLinux uses a "symbolic link" (ln -s ...) in places to create references on the non-persistent /var/ tree that points to persistent storage found in the /mnt/kd/ tree as well as to the read-only AstLinux ext2 flash image.

These "symlinks" can be confusing to follow at first, for example following your "custom-sounds" case ...
--
pbx ~ # ls -l /var/lib/asterisk/sounds/custom-sounds
lrwxrwxrwx    1 root     root            38 Aug 27 16:40 /var/lib/asterisk/sounds/custom-sounds -> /var/tmp/asterisk/sounds/custom-sounds

pbx ~ # ls -l /var/tmp/asterisk/sounds/custom-sounds
lrwxrwxrwx    1 root     root            21 Aug 27 16:41 /var/tmp/asterisk/sounds/custom-sounds -> /mnt/kd/custom-sounds
--
Note the " -> " indicates a symlink reference.

Remember you should only be adding/editing files in the /mnt/kd/ tree path, though if you know a symlink points to the /mnt/kd/ tree path you can use the symlink name as a convenience if you wish.

Lonnie


On Aug 27, 2017, at 10:34 PM, Cody Alderson <ald...@gm...> wrote:

> Lonnie,
> 
> It works. I did as you said, and then I put the files back in /var/lib/asterisk/custom-sounds that did already exist as that is where I kept putting them. The /mnt/kd/custom-sounds did not exist before I did the mkdir. I checked. I made the directory, then I used the CLI to mount a flash drive and put the ulaw files back in /var/lib/asterisk/sounds/custom-sounds. They remained this time after a reboot. I don't understand persistent directories yet. When I use winSCP to look at the tree structure, I see the same directories repeated under /stat and /tmp for /var/lib/asterisk/sounds/custom-sounds and other directories. I am not fully understanding Linux directories yet. The book I'm reading is not great at explaining Linux either.
> 
> Thank you for the help!
> 
> 
> -Cody
> 
> On Sun, Aug 27, 2017 at 9:54 AM, Lonnie Abelbeck <li...@lo...> wrote:
> Hi Cody,
> 
> Yes, AstLinux uses "tmpfs" (RAM based) file storage for some paths.  The official persistent storage path is /mnt/kd/ and below is always saved across reboots.
> 
> For the special case of Asterisk sound files, we offer a useful symlink to a /mnt/kd/custom-sounds directory if it exists.
> 
> This is what I suggest you do from the CLI ...
> --
> mkdir /mnt/kd/custom-sounds
> 
> service asterisk stop
> service asterisk init
> --
> 
> With this the /var/lib/asterisk/sounds/custom-sounds path points to /mnt/kd/custom-sounds, so if you had a sound file /mnt/kd/custom-sounds/greeting.ulaw you could reference it via the Asterisk dialplan as "custom-sounds/greeting.ulaw".
> 
> You can also add additional directories in /mnt/kd/custom-sounds/ say /mnt/kd/custom-sounds/tts/ as such you could reference sound files as "custom-sounds/tts/greeting.ulaw".
> 
> Clear ?
> 
> Lonnie
> 
> 
> 
> On Aug 27, 2017, at 8:28 AM, Cody Alderson <ald...@gm...> wrote:
> 
> > Hi,
> >
> > I used TTS to make some custom prompts for an extension that plays humorous TTS files based on day and time. When I reboot the thin client Astlinux is running on, the directory remains but the files are deleted. I used the CLI to make the directory and move the files I made into it. Everything works fine until a reboot, and then those files in that directory are gone. Would someone please advise me as to what I am doing wrong? Keep in mind that I am an Asterisk and Linux novice.
> >
> > Thank you!
> >
> > -Cody Alderson

Re: [Astlinux-users] My Custom TTS Ulaw Sound Files Delete on Reboot of Thin Client Running Astlinux

[Cody A. <ald...@gm...>]

Lonnie,

It works. I did as you said, and then I put the files back in
/var/lib/asterisk/custom-sounds that did already exist as that is where I
kept putting them. The /mnt/kd/custom-sounds did not exist before I did the
mkdir. I checked. I made the directory, then I used the CLI to mount a
flash drive and put the ulaw files back in
/var/lib/asterisk/sounds/custom-sounds. They remained this time after a
reboot. I don't understand persistent directories yet. When I use winSCP to
look at the tree structure, I see the same directories repeated under /stat
and /tmp for /var/lib/asterisk/sounds/custom-sounds and other directories.
I am not fully understanding Linux directories yet. The book I'm reading is
not great at explaining Linux either.

Thank you for the help!


-Cody

On Sun, Aug 27, 2017 at 9:54 AM, Lonnie Abelbeck <li...@lo...>
wrote:

> Hi Cody,
>
> Yes, AstLinux uses "tmpfs" (RAM based) file storage for some paths.  The
> official persistent storage path is /mnt/kd/ and below is always saved
> across reboots.
>
> For the special case of Asterisk sound files, we offer a useful symlink to
> a /mnt/kd/custom-sounds directory if it exists.
>
> This is what I suggest you do from the CLI ...
> --
> mkdir /mnt/kd/custom-sounds
>
> service asterisk stop
> service asterisk init
> --
>
> With this the /var/lib/asterisk/sounds/custom-sounds path points to
> /mnt/kd/custom-sounds, so if you had a sound file /mnt/kd/custom-sounds/greeting.ulaw
> you could reference it via the Asterisk dialplan as
> "custom-sounds/greeting.ulaw".
>
> You can also add additional directories in /mnt/kd/custom-sounds/ say
> /mnt/kd/custom-sounds/tts/ as such you could reference sound files as
> "custom-sounds/tts/greeting.ulaw".
>
> Clear ?
>
> Lonnie
>
>
>
> On Aug 27, 2017, at 8:28 AM, Cody Alderson <ald...@gm...> wrote:
>
> > Hi,
> >
> > I used TTS to make some custom prompts for an extension that plays
> humorous TTS files based on day and time. When I reboot the thin client
> Astlinux is running on, the directory remains but the files are deleted. I
> used the CLI to make the directory and move the files I made into it.
> Everything works fine until a reboot, and then those files in that
> directory are gone. Would someone please advise me as to what I am doing
> wrong? Keep in mind that I am an Asterisk and Linux novice.
> >
> > Thank you!
> >
> > -Cody Alderson
>
>
>
>
> ------------------------------------------------------------
> ------------------
> Check out the vibrant tech community on one of the world's most
> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>

Re: [Astlinux-users] My Custom TTS Ulaw Sound Files Delete on Reboot of Thin Client Running Astlinux

[Cody A. <ald...@gm...>]

Lonnie,

Yes, it is clear. I will get back to you after I try it. Thank you!

-Cody

On Sun, Aug 27, 2017 at 9:54 AM, Lonnie Abelbeck <li...@lo...>
wrote:

> Hi Cody,
>
> Yes, AstLinux uses "tmpfs" (RAM based) file storage for some paths.  The
> official persistent storage path is /mnt/kd/ and below is always saved
> across reboots.
>
> For the special case of Asterisk sound files, we offer a useful symlink to
> a /mnt/kd/custom-sounds directory if it exists.
>
> This is what I suggest you do from the CLI ...
> --
> mkdir /mnt/kd/custom-sounds
>
> service asterisk stop
> service asterisk init
> --
>
> With this the /var/lib/asterisk/sounds/custom-sounds path points to
> /mnt/kd/custom-sounds, so if you had a sound file /mnt/kd/custom-sounds/greeting.ulaw
> you could reference it via the Asterisk dialplan as
> "custom-sounds/greeting.ulaw".
>
> You can also add additional directories in /mnt/kd/custom-sounds/ say
> /mnt/kd/custom-sounds/tts/ as such you could reference sound files as
> "custom-sounds/tts/greeting.ulaw".
>
> Clear ?
>
> Lonnie
>
>
>
> On Aug 27, 2017, at 8:28 AM, Cody Alderson <ald...@gm...> wrote:
>
> > Hi,
> >
> > I used TTS to make some custom prompts for an extension that plays
> humorous TTS files based on day and time. When I reboot the thin client
> Astlinux is running on, the directory remains but the files are deleted. I
> used the CLI to make the directory and move the files I made into it.
> Everything works fine until a reboot, and then those files in that
> directory are gone. Would someone please advise me as to what I am doing
> wrong? Keep in mind that I am an Asterisk and Linux novice.
> >
> > Thank you!
> >
> > -Cody Alderson
>
>
>
>
> ------------------------------------------------------------
> ------------------
> Check out the vibrant tech community on one of the world's most
> engaging tech sites, Slashdot.org! http://sdm.link/slashdot
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pa...@kr....
>

[Astlinux-users] cant log back into Astlinux Management GUI

[Roberto R. <rri...@gm...>]

Hi ALL,

Before you read about my issue I want to let everyone know that I have been
tinkering with Asterisk for about 5-6 month. So Im a total new guy.

With that being said, Im hoping someone can point me in the right direction
or you could tell me what you did to overcome this issue if you have
experienced it before.

About a month ago I decided to learn how to use the Astlinux Management GUI
to do some testing with hard and soft phone (Im in the call center business
Im not an IT person) learning from Youtube videos how to set up dial plans
ect. using Centos 6 and Linux. Everything was going well. I was able to set
up a small Lab at home for testing and learning.

Fast forward to Astlinux Management Software.
I learned how to log in using the ip address of my appliance, PC Engines
Alix Board. Everything was going well until one day I noticed a message
indicating that I had run out of memory....Fine, I rebooted and logged back
in as normal and everything was good. I was able to make inbound and
outbound calls and dial extension to extension.

I also noticed a ton of unauthorized registration attempts but they were
all rejected because of "wrong password". I was sure they were brute force
attempts to hack into the software.

Anyway, one day last week I noticed the same memory issue and so I rebooted
as had done before but this time I could not log back in (
https://xxx.xxx.x.xx).
This is the screen that comes up:

This site can’t be reached

*192.168.1.25* took too long to respond.

Try:

   - Checking the connection
   - Checking the proxy and the firewall
   - Running Windows Network Diagnostics

ERR_CONNECTION_TIMED_OUT

*After running windows network diagnostics I get this report that states:*

resource (MY PBX IP address) is online but isn't responding to connection
attempts.

The remote computer isn't responding to connections on port 443, possibly
due to firewall or security policy settings, or because it might be
temporarily unavailable. windows could not find any problem with the
firewall on your computer.

contact the service provider or owner of the remote *system* for further
assistance, or try again later.

In my research I have found some similar issues on youtube and Google but I
continue to get the same message as above.
The only post I found was on a Asterisk forum that basically stated that a
hacker could deploy a script that would disable the user from logging back
in. Im starting to believe this is what happened in my situation. How could
I get back into the software to continue learning? From what I have read
online the upgrade to Astlinux 1.3 would still allow me continue using the
software with what I was using to learn. (Asterisk 11 and Alix board
2f6).It was working just fine as I had read it would.
Any help would be greatly appreciated.

Thanks All!

Re: [Astlinux-users] My Custom TTS Ulaw Sound Files Delete on Reboot of Thin Client Running Astlinux

[Lonnie A. <li...@lo...>]

Hi Cody,

Yes, AstLinux uses "tmpfs" (RAM based) file storage for some paths.  The official persistent storage path is /mnt/kd/ and below is always saved across reboots.

For the special case of Asterisk sound files, we offer a useful symlink to a /mnt/kd/custom-sounds directory if it exists.

This is what I suggest you do from the CLI ...
--
mkdir /mnt/kd/custom-sounds

service asterisk stop
service asterisk init
--

With this the /var/lib/asterisk/sounds/custom-sounds path points to /mnt/kd/custom-sounds, so if you had a sound file /mnt/kd/custom-sounds/greeting.ulaw you could reference it via the Asterisk dialplan as "custom-sounds/greeting.ulaw".

You can also add additional directories in /mnt/kd/custom-sounds/ say /mnt/kd/custom-sounds/tts/ as such you could reference sound files as "custom-sounds/tts/greeting.ulaw".

Clear ?

Lonnie



On Aug 27, 2017, at 8:28 AM, Cody Alderson <ald...@gm...> wrote:

> Hi,
> 
> I used TTS to make some custom prompts for an extension that plays humorous TTS files based on day and time. When I reboot the thin client Astlinux is running on, the directory remains but the files are deleted. I used the CLI to make the directory and move the files I made into it. Everything works fine until a reboot, and then those files in that directory are gone. Would someone please advise me as to what I am doing wrong? Keep in mind that I am an Asterisk and Linux novice.
> 
> Thank you!
> 
> -Cody Alderson

[Astlinux-users] My Custom TTS Ulaw Sound Files Delete on Reboot of Thin Client Running Astlinux

[Cody A. <ald...@gm...>]

Hi,

I used TTS to make some custom prompts for an extension that plays humorous
TTS files based on day and time. When I reboot the thin client Astlinux is
running on, the directory remains but the files are deleted. I used the CLI
to make the directory and move the files I made into it. Everything works
fine until a reboot, and then those files in that directory are gone. Would
someone please advise me as to what I am doing wrong? Keep in mind that I
am an Asterisk and Linux novice.

Thank you!

-Cody Alderson

Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[Tim T. <tt...@z-...>]

Found the problem!  In voicemail.conf, there is a setting called ‘exitcontext’  It was set to ‘vm-operator’.   I found this on the voip-info.org site:

exitcontext

Optional context to drop the user into after he/she has pressed * or 0 to exit voicemail. If not set, pressing * or 0 will return the caller to the last context they were in before being sent to voicemail (assuming that context has a 'a' or 'o' extension).

 

 

After commenting out this setting, pressing ‘ * ‘ will drop me right into the proper mailbox asking for a password.

 

Thanks again for all your help and ideas on this.

 

From: The Cadillac Kid via Astlinux-users [mailto:ast...@li...] 
Sent: Wednesday, August 23, 2017 7:51 PM
To: AstLinux Users Mailing List
Cc: The Cadillac Kid
Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

 

it definitely works while playing the greeting... 

 

 

 

    -- Executing [s@macro-ringphone:207] CELGenUserEvent("SIP/4999-00000001", "VMSCOVER,1503518585.1,4001") in new stack
    -- Executing [s@macro-ringphone:208] VoiceMail("SIP/4999-00000001", "4001@default,u") in new stack
       > 0x9af5428 -- Probation passed - setting RTP source address to 172.16.37.1:50478
    -- <SIP/4999-00000001> Playing 'vm-theperson.ulaw' (language 'en')
    -- <SIP/4999-00000001> Playing 'digits/4.ulaw' (language 'en')
    -- <SIP/4999-00000001> Playing 'digits/0.ulaw' (language 'en')
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4214 __ast_read: DTMF begin '*' received on SIP/4999-00000001
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4218 __ast_read: DTMF begin ignored '*' on SIP/4999-00000001
    -- <SIP/4999-00000001> Playing 'digits/0.ulaw' (language 'en')
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4128 __ast_read: DTMF end '*' received on SIP/4999-00000001, duration 160 ms
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4198 __ast_read: DTMF end passthrough '*' on SIP/4999-00000001
    -- Executing [a@macro-ringphone:1] Set("SIP/4999-00000001", "mailboxnum=4001") in new stack
    -- Executing [a@macro-ringphone:2] NoOp("SIP/4999-00000001", "called by:featureset-dial") in new stack
    -- Executing [a@macro-ringphone:3] Set("SIP/4999-00000001", "boxpass=7623") in new stack
    -- Executing [a@macro-ringphone:4] Set("SIP/4999-00000001", "adminpro=admin-") in new stack
    -- Executing [a@macro-ringphone:5] GotoIf("SIP/4999-00000001", "0?voicemenu-checkvm,s,logmeout") in new stack
    -- Executing [a@macro-ringphone:6] Goto("SIP/4999-00000001", "voicemail-login,s,starlog") in new stack
    -- Goto (voicemail-login,s,7)

 

 

make sure that your a extension is recognized by asterisk..  do a dialplan show of your context..  below is an example of mine where the 'a' extension shows..  

I looked through the source code of 11.20 and didnt see nay config options that need set to enable it..  be sure you did a dialplan reload after you make changes to your contexts (or are you running realtime?) 

 

-Christopher

 

VM*CLI> dialplan show macro-ringphone
[ Context 'macro-ringphone' created by 'pbx_config' ]
  'a' =>            1. Set(mailboxnum=${cidreceiver})             [pbx_config]
                    2. NoOp(called by:${MACRO_CONTEXT})           [pbx_config]
                    3. Set(boxpass=${DB(vmpass/${mailboxnum})})   [pbx_config]
                    4. Set(adminpro=${IF($[$["${DB(active/${mailboxnum})}" != "yes"] & $["${DB(active/${mailboxnum})}" != "no"]]?admin-)}) [pbx_config]
                    5. GotoIf($[$["${mailboxnum}" = "${boxpass}"] || ["${boxpass}" = "${DEFAULT_VM_PASSCODE}"]]?voicemenu-checkvm,s,logmeout) [pbx_config]
                    6. Goto(voicemail-login,s,starlog)            [pbx_config]
                    7. Hangup()                                   [pbx_config]

 

  _____  

From: Lonnie Abelbeck <li...@lo...>
To: AstLinux Users Mailing List <ast...@li...> 
Sent: Wednesday, August 23, 2017 4:32 PM
Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

 

Tim,

 

For testing you might try also adding the 'd' option to VoiceMail()

--

d - Accept digits for a new extension in context c, if played during the greeting. Context defaults to the current context.

--

try "1" first then "*" .

 

https://wiki.asterisk.org/wiki/display/AST/Application_VoiceMail

 

>From reading the docs I'm not sure if

--

* - Jump to the 'a' extension in the current dialplan context.

--

works while playing the greeting.

 

Lonnie

 

 

On Aug 23, 2017, at 3:09 PM, Tim Turpin <tt...@z-...> wrote:

 

> I pressed ‘*’ twice while listening to my unavailable greeting, nothing happened.

>  

> I believe Asterisk is doing nothing with the ‘*’:

>  

>  

>    -- Executing [9373506524@inbound:5] VoiceMail("SIP/voipms-00000046", "9373506524,u") in new stack

> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: app_voicemail.c:6413 leave_voicemail: Before find_user

> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: channel.c:5414 set_format: Set channel SIP/voipms-00000046 to write format slin

> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3446 ast_rtp_write: Ooh, format changed from unknown to ulaw

> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3481 ast_rtp_write: Created smoother: format: ulaw ms: 20 len: 160

> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3343 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x2addf4026628'

> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second

>    -- <SIP/voipms-00000046> Playing '/var/spool/asterisk/voicemail/default/9373506524/unavail.slin' (language 'en')

> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:4333 ast_rtp_read: 0x2addf402b830 -- Probation learning mode pass with source address 72.9.246.170:13730

> [Aug 23 15:50:37] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730

> [Aug 23 15:50:37] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730

> [Aug 23 15:50:38] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730

> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730

> [Aug 23 15:50:39] DEBUG[482]: chan_sip.c:4285 __sip_autodestruct: Auto destroying SIP dialog '0190242c0812028377b2281e2df47b3b@72.9.246.170:5060'

> [Aug 23 15:50:39] DEBUG[482]: chan_sip.c:6379 sip_pvt_dtor: Destroying SIP dialog 0190242c0812028377b2281e2df47b3b@72.9.246.170:5060

> [Aug 23 15:50:39] DEBUG[482]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x2addf4002d98'

> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (58 requested / 58 actual) timer ticks per second

> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second

> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second

> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second

> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:5414 set_format: Set channel SIP/voipms-00000046 to write format ulaw

> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second

>  

>    -- <SIP/voipms-00000046> Playing 'vm-intro.ulaw' (language 'en')

> [Aug 23 15:50:40] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730

> [Aug 23 15:50:40] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730

>  

> [Aug 23 15:50:42] DEBUG[482]: chan_sip.c:9057 find_call: = Looking for  Call ID: 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060 (Checking From) --From tag as2b5c0e97 --To-tag as7ac59689

> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:28533 handle_incoming: **** Received BYE (8) - Command in SIP BYE

> [Aug 23 15:50:42] DEBUG[482][C-00000052]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '72.9.246.170:5060' into...

> [Aug 23 15:50:42] DEBUG[482][C-00000052]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '72.9.246.170' and port '5060'.

> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:3387 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060

> [Aug 23 15:50:42] DEBUG[482][C-00000052]: res_rtp_asterisk.c:4755 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2addf4026628'

> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:29442 stop_session_timer: Session timer stopped: 1 - 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060

> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:27149 handle_request_bye: Received bye, issuing owner hangup

> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:3731 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 72.9.246.170:5060

> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: pbx.c:6789 __ast_pbx_run: Spawn extension (inbound,9373506524,5) exited non-zero on 'SIP/voipms-00000046'

>  == Spawn extension (inbound, 9373506524, 5) exited non-zero on 'SIP/voipms-00000046'

> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:2662 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/voipms-00000046'

> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: pbx.c:2111 new_find_extension: return at end of func

> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second

> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second

> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:2841 ast_hangup: Hanging up channel 'SIP/voipms-00000046'

> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: chan_sip.c:6929 sip_hangup: Hangup call SIP/voipms-00000046, SIP callid 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060

> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:4755 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2addf4026628'

> [Aug 23 15:50:42] DEBUG[438]: devicestate.c:345 _ast_device_state: No provider found, checking channel drivers for SIP - voipms

> [Aug 23 15:50:42] DEBUG[438]: chan_sip.c:29982 sip_devicestate: Checking device state for peer voipms

> [Aug 23 15:50:42] DEBUG[438]: devicestate.c:477 do_state_change: Changing state for SIP/voipms - state 1 (Not in use)

> [Aug 23 15:50:42] DEBUG[438]: devicestate.c:452 devstate_event: device 'SIP/voipms' state '1'

> [Aug 23 15:50:42] DEBUG[509]: app_queue.c:1924 handle_statechange: Device 'SIP/voipms' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

>  

> It doesn’t appear to be taking any action at all.  The system continues to record the message and delivers out to email.  Is it possible that the ‘a’ extension is broken?

>  

>  

> From: David Kerr [mailto:da...@ke...] 

> Sent: Wednesday, August 23, 2017 2:00 PM

> To: AstLinux Users Mailing List

> Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

>  

> That tells you that Asterisk is detecting the tone.  Doesn't tell you what it is doing with it... so you still need to trace dialplan execution (turn off debug, leave verbose on) to see what action it is taking on the tone.

>  

> David

>  

> On Wed, Aug 23, 2017 at 12:13 PM, Tim Turpin <tt...@z-...> wrote:

> I won’t copy in the entire session (way too much info), but here’s the result  of my pressing *,*,1,2,3,4,5,6,#.  It looks as though Asterisk is seeing the DTMF.

>  

>  

> [Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:12772

> 

> [Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:12772

> 

> [Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:12772

> 

> [Aug 23 11:58:48] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:12772

> 

> [Aug 23 11:58:48] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 49 (1), at 72.9.246.170:12772

> 

> [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 49 (1), at 72.9.246.170:12772

> 

> [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 50 (2), at 72.9.246.170:12772

> 

> [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 50 (2), at 72.9.246.170:12772

> 

> [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 51 (3), at 72.9.246.170:12772

> 

> [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 51 (3), at 72.9.246.170:12772

> 

> [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 52 (4), at 72.9.246.170:12772

> 

> [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 52 (4), at 72.9.246.170:12772

> 

> [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 53 (5), at 72.9.246.170:12772

> 

> [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 53 (5), at 72.9.246.170:12772

> 

> [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 54 (6), at 72.9.246.170:12772

> 

> [Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 54 (6), at 72.9.246.170:12772

> 

> [Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 35 (#), at 72.9.246.170:12772

> 

> [Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 35 (#), at 72.9.246.170:12772

> 

>  

>  

>  

> From: David Kerr [mailto:da...@ke...] 

> Sent: Wednesday, August 23, 2017 11:05 AM

> To: AstLinux Users Mailing List

> Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

>  

> Check that the * key is not being captured for some other purpose (grep into other .conf files).  Check that you can match the * key outside of voicemail... use WaitExten() and validate that your dialplan sees that.  You can also go into the asterisk console ("asterisk -r") and turn on verbose and debug... e.g. "core set verbose 999" and "core set debug 999" and watch in the console.... make sure that logger.conf has a line that says "console => notice,warning,error,debug,verbose" else you might not get the debug and verbose messages into your console.

>  

> David

>  

> On Wed, Aug 23, 2017 at 10:53 AM, Tim Turpin <tt...@z-...> wrote:

> If I change my config to direct the call to VoiceMailMain(), I can log in

> with DTMF digits, so I know the carrier is passing tones. And Asterisk is

> recognizing them.

> Thanks.

> 

> -----Original Message-----

> From: Lonnie Abelbeck [mailto:li...@lo...]

> Sent: Wednesday, August 23, 2017 10:51 AM

> To: AstLinux Users Mailing List

> Subject: Re: [Astlinux-users] Question about setting up AstLinux as

> voicemail server

> 

> Tim,

> 

> Make sure in your sip.conf for your inbound provider the setting for

> "dtmfmode" matches what your provider requires, Asterisk defaults to rfc2833

> .

> 

> Lonnie

> 

> 

> On Aug 23, 2017, at 9:20 AM, Tim Turpin <tt...@z-...> wrote:

> 

> > Getting closer, I think.

> >

> > I'm starting to wonder if the DTMF '*' is being recognized at all.  Now

> the caller is dropped into the proper mailbox, but pressing '*' does

> nothing.

> > Here's extensions.conf:

> >

> > [inbound]

> >

> > exten => _NXXNXXXXXX,1,Answer

> > exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call)

> > exten => _NXXNXXXXXX,n,Set(boxnumber=${EXTEN})    ; set a variable for box

> number

> > exten => _NXXNXXXXXX,n,NoOp(${boxnumber})        ;  test for variable

> > exten => _NXXNXXXXXX,n,Voicemail(${boxnumber})

> > exten => _NXXNXXXXXX,n,Hangup

> > exten => a,1,VoiceMailMain(${boxnumber})      ; user dialed * in

> greeting. send them to their mailbox

> > exten => a,n,Hangup

> >

> >

> > Here's the response when calling the DID number 9373506524:

> >

> > Connected to Asterisk 11.25.1 currently running on SST (pid = 415)

> >  == Using SIP RTP CoS mark 5

> >    -- Executing [9373506524@inbound:1] Answer("SIP/voipms-00000037", "")

> in new stack

> >    -- Executing [9373506524@inbound:2] NoOp("SIP/voipms-00000037",

> "inbound-phone-call") in new stack

> >    -- Executing [9373506524@inbound:3] Set("SIP/voipms-00000037",

> "boxnumber=9373506524") in new stack

> >    -- Executing [9373506524@inbound:4] NoOp("SIP/voipms-00000037",

> "9373506524") in new stack

> >    -- Executing [9373506524@inbound:5] VoiceMail("SIP/voipms-00000037",

> "9373506524") in new stack

> >    -- <SIP/voipms-00000037> Playing

> '/var/spool/asterisk/voicemail/default/9373506524/temp.slin' (language 'en')

> >    -- <SIP/voipms-00000037> Playing 'vm-intro.ulaw' (language 'en')

> >    -- <SIP/voipms-00000037> Playing 'beep.ulaw' (language 'en')

> >    -- Recording the message

> >    -- x=0, open writing:

> > /var/spool/asterisk/voicemail/default/9373506524/tmp/2u8Hzw format:

> > wav, 0x2addfc001798

> >

> > Is there any setting that would not allow the '*' to be recognized during

> the greeting?

> >

> >

> >

> >

> > From: The Cadillac Kid via Astlinux-users

> > [mailto:ast...@li...]

> > Sent: Wednesday, August 23, 2017 8:32 AM

> > To: AstLinux Users Mailing List

> > Cc: The Cadillac Kid

> > Subject: Re: [Astlinux-users] Question about setting up AstLinux as

> > voicemail server

> >

> > set a variable first... the issue is that ${EXTEN} changes to 'a' when you

> * out...  ${EXTEN} is the current extension you are workign with and you

> want to go to the original dialed extension.

> >

> > [inbound]

> > exten => _NXXNXXXXXX,1,Answer

> > exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call)

> > ; set a variable for box number

> > exten => _NXXNXXXXXX,n,Set(boxumber=${EXTEN})

> >

> > exten => _NXXNXXXXXX,n,Voicemail(${boxnumber})

> > ;exten => _NXXNXXXXXX,n,VoiceMailMain(${EXTEN})

> > exten = > _NXXNXXXXXX,n,Hangup

> >

> > ; user dialed * in greeting. send them to their mailbox

> >

> > exten => a, 1, VoicemailMain(${boxnumber}) exten => a,n, Hangup

> >

> >

> >

> > -Christopher

> >

> >

> > From: Tim Turpin <tt...@z-...>

> > To: 'AstLinux Users Mailing List'

> > <ast...@li...>

> > Sent: Wednesday, August 23, 2017 8:14 AM

> > Subject: Re: [Astlinux-users] Question about setting up AstLinux as

> > voicemail server

> >

> > This appears to possibly work for one mailbox user.  We have a couple

> thousand users, all dialing in via DID, and the process needs to be the same

> for all users.  My current extensions.conf looks like this:

> >

> > [inbound]

> > exten => _NXXNXXXXXX,1,Answer

> > exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call)

> > exten => _NXXNXXXXXX,n,Voicemail(${EXTEN}) ;exten =>

> > _NXXNXXXXXX,n,VoiceMailMain(${EXTEN})

> > ;exten => a, 1, VoicemailMain(${EXTEN})

> >

> > I've played with the 'a' extension in different formats, but can't seem to

> make it work.  In the current configuration, when a caller dials in, it

> plays the greeting for that particular mailbox.  If I comment out the third

> line and un-comment the fourth, the caller drops into their box with the

> ability to log in.  I can't figure out how to utilize the 'a' extension to

> allow the user to press '*' to login while listening to his greeting (the

> fifth line).

> >

> > I'm using information about the 'a' extension from the following sites:

> >

> > From ' https://www.voip-info.org/wiki-asterisk+standard+extensions1  <https://www.voip-info.org/wiki-asterisk+standard+extensions1> ':

> > a: Called when user presses '*' during a voicemail greeting

> > h: Hangup extension

> > i: invalid extension

> > o: Operator extension, used for operator exit by pressing zero in

> > voicemail

> > s: Start extension in context

> > t: Timeout extension

> > T: AbsoluteTimeout() extension

> > Also, from ' https://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail  <https://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail> ':

> > Also. during the prompt if the caller presses:

> > '*' - the call jumps to extension 'a' in the current voicemail context.

> > Example:

> > Exten => a, 1, VoicemailMain(@default) Exten => a, 2, Hangup Being a

> > novice at Asterisk, I have to assume that I'm not following the proper

> coding format, or I'm not applying the 'a' extension properly.  From what I

> have read on these two web pages, I think that this is the application to

> use, but I'm just not applying it properly.

> >

> > From: David Kerr [mailto:da...@ke...]

> > Sent: Tuesday, August 22, 2017 5:30 PM

> > To: AstLinux Users Mailing List

> > Subject: Re: [Astlinux-users] Question about setting up AstLinux as

> > voicemail server

> >

> > Tim,

> >  You are going to want to use the Background() app to play the greeting

> with the WaitExten() app to wait for a keypress (if they wait til the very

> end of the greeting before pressing) and then the Authenticate() app to get

> a PIN to proceed to whatever action is permitted.  Something like this

> (untested but should be close enough)...

> >

> > [leavemessage]

> > exten = s,1,NoOp(voicemail)

> >  same = n,Ringing()

> >  same = n,Wait(2)

> >  same = n,Answer()

> >  same = n(start),Set(TIMEOUT(response)=1)  same =

> > n,Set(TIMEOUT(digit)=1)  same = n,Background(record/NoAnswer) ; my

> > custom message, press 1 or wait to leave a msg  same = n,WaitExten(1)

> > exten = 1,1,Voicemail(101,us) ; caller pressed 1  same = n,NoOp(Back

> > from voicemail)  same = n,Hangup() exten =

> > _[*],1,VoiceMailMain(101,sa(0)) ; caller pressed *  same = n,NoOp(Back

> > from voicemailmain)  same = n,Hangup() exten = t,1,Voicemail(101,us) ;

> > timeout, leave a message. could GoTo(1,1)  same = n,NoOp(Back from

> > voicemail)  same = n,Hangup() exten = i,1,Playback(pbx-invalid) ;

> > standard invalid key pressed msg.

> >  same = n,Goto(s,start)

> > exten = h,1,Hangup()

> >

> > David

> >

> >

> >

> > On Tue, Aug 22, 2017 at 3:04 PM, Tim Turpin <tt...@z-...> wrote:

> > Thank you for the fast reply.

> >

> > I loaded up the AstLinux last week. I've been able to figure out most

> > of what I need, except for a way to route incoming DID calls to

> > voicemail, allowing the caller to be able to press '*' while hearing

> > the mailbox greeting and then be handed off to 'VoiceMailMain()' to log

> into their box.

> > If '*' isn't pressed, the caller would just drop into the mailbox to

> > leave a message.

> >

> > It seems like it should be easy to set up, but it's really kicking my

> > butt right now, and I'm just trying to determine my best avenue for

> > assistance in figuring this out.  I'll try the Asterisk forums and see

> > if they can offer any help.

> >

> > Thanks again.

> >

> > Tim

> >

> >

> >

> > -----Original Message-----

> > From: Lonnie Abelbeck [mailto:li...@lo...]

> > Sent: Tuesday, August 22, 2017 2:31 PM

> > To: AstLinux Users Mailing List

> > Subject: Re: [Astlinux-users] Question about setting up AstLinux as

> > voicemail server

> >

> >

> > On Aug 22, 2017, at 11:49 AM, Tim Turpin <tt...@z-...> wrote:

> >

> > > I'm new to the Asterisk world, and I'm trying to use AstLinux to

> > > replicate

> > an existing voicemail environment, and I have several configuration

> > questions.  Is this the proper forum for these questions, or do I send

> > the questions somewhere else?

> > >

> > > Thanks.

> > > Tim.

> >

> > Hi Tim,

> >

> > First, using AstLinux as a dedicated voicemail server, using a small

> > x86 appliance and SSD storage or Virtual Machine Guest is a good approach.

> >

> > This mailing list is mostly dedicated to AstLinux Project specific

> > questions, Asterisk voicemail.conf, sip.conf and extensions.conf

> > configurations are best asked in the Asterisk support groups.  If you

> > have things all but working and have reached a brick wall using

> > AstLinux ... you can give this list a try.

> >

> > Keep in mind that using AstLinux, you will be required to generate the

> > base extensions.conf text file for yourself, AstLinux has a basic web

> > interface and "Users" tab that can help manage your voicemail users.

> > As a starting point you might spin-up the "Guest VM x86-64bit (Video

> > Console)" Install ISO in a virtual machine to give you a playground to

> > test before purchasing any hardware.

> >

> > Alternatively, if coding a extensions.conf is not your cup-of-tea you

> > might query this mailing list for off-line consulting help.

> >

> > Here is a reference to give you the flavor of the configuration ...

> >

> > Configuring Voice Mail Boxes

> > https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxe

> > s

> >

> > Lonnie

> >

 

 

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Re: [Astlinux-users] Question about setting up AstLinux as voicemail server

[The C. K. <eld...@ya...>]

it definitely works while playing the greeting... 



    -- Executing [s@macro-ringphone:207] CELGenUserEvent("SIP/4999-00000001", "VMSCOVER,1503518585.1,4001") in new stack
    -- Executing [s@macro-ringphone:208] VoiceMail("SIP/4999-00000001", "4001@default,u") in new stack
       > 0x9af5428 -- Probation passed - setting RTP source address to 172.16.37.1:50478
    -- <SIP/4999-00000001> Playing 'vm-theperson.ulaw' (language 'en')
    -- <SIP/4999-00000001> Playing 'digits/4.ulaw' (language 'en')
    -- <SIP/4999-00000001> Playing 'digits/0.ulaw' (language 'en')
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4214 __ast_read: DTMF begin '*' received on SIP/4999-00000001
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4218 __ast_read: DTMF begin ignored '*' on SIP/4999-00000001
    -- <SIP/4999-00000001> Playing 'digits/0.ulaw' (language 'en')
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4128 __ast_read: DTMF end '*' received on SIP/4999-00000001, duration 160 ms
[2017-08-23 16:03:08] DTMF[3310][C-00000001]: channel.c:4198 __ast_read: DTMF end passthrough '*' on SIP/4999-00000001
    -- Executing [a@macro-ringphone:1] Set("SIP/4999-00000001", "mailboxnum=4001") in new stack
    -- Executing [a@macro-ringphone:2] NoOp("SIP/4999-00000001", "called by:featureset-dial") in new stack
    -- Executing [a@macro-ringphone:3] Set("SIP/4999-00000001", "boxpass=7623") in new stack
    -- Executing [a@macro-ringphone:4] Set("SIP/4999-00000001", "adminpro=admin-") in new stack
    -- Executing [a@macro-ringphone:5] GotoIf("SIP/4999-00000001", "0?voicemenu-checkvm,s,logmeout") in new stack
    -- Executing [a@macro-ringphone:6] Goto("SIP/4999-00000001", "voicemail-login,s,starlog") in new stack
    -- Goto (voicemail-login,s,7)

make sure that your a extension is recognized by asterisk..  do a dialplan show of your context..  below is an example of mine where the 'a' extension shows..  
I looked through the source code of 11.20 and didnt see nay config options that need set to enable it..  be sure you did a dialplan reload after you make changes to your contexts (or are you running realtime?) 

-Christopher

VM*CLI> dialplan show macro-ringphone
[ Context 'macro-ringphone' created by 'pbx_config' ]
  'a' =>            1. Set(mailboxnum=${cidreceiver})             [pbx_config]
                    2. NoOp(called by:${MACRO_CONTEXT})           [pbx_config]
                    3. Set(boxpass=${DB(vmpass/${mailboxnum})})   [pbx_config]
                    4. Set(adminpro=${IF($[$["${DB(active/${mailboxnum})}" != "yes"] & $["${DB(active/${mailboxnum})}" != "no"]]?admin-)}) [pbx_config]
                    5. GotoIf($[$["${mailboxnum}" = "${boxpass}"] || ["${boxpass}" = "${DEFAULT_VM_PASSCODE}"]]?voicemenu-checkvm,s,logmeout) [pbx_config]
                    6. Goto(voicemail-login,s,starlog)            [pbx_config]
                    7. Hangup()                                   [pbx_config]


      From: Lonnie Abelbeck <li...@lo...>
 To: AstLinux Users Mailing List <ast...@li...> 
 Sent: Wednesday, August 23, 2017 4:32 PM
 Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail server
   
Tim,

For testing you might try also adding the 'd' option to VoiceMail()
--
d - Accept digits for a new extension in context c, if played during the greeting. Context defaults to the current context.
--
try "1" first then "*" .

https://wiki.asterisk.org/wiki/display/AST/Application_VoiceMail

>From reading the docs I'm not sure if
--
* - Jump to the 'a' extension in the current dialplan context.
--
works while playing the greeting.

Lonnie


On Aug 23, 2017, at 3:09 PM, Tim Turpin <tt...@z-...> wrote:

> I pressed ‘*’ twice while listening to my unavailable greeting, nothing happened.
>  
> I believe Asterisk is doing nothing with the ‘*’:
>  
>  
>    -- Executing [9373506524@inbound:5] VoiceMail("SIP/voipms-00000046", "9373506524,u") in new stack
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: app_voicemail.c:6413 leave_voicemail: Before find_user
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: channel.c:5414 set_format: Set channel SIP/voipms-00000046 to write format slin
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3446 ast_rtp_write: Ooh, format changed from unknown to ulaw
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3481 ast_rtp_write: Created smoother: format: ulaw ms: 20 len: 160
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3343 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x2addf4026628'
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
>    -- <SIP/voipms-00000046> Playing '/var/spool/asterisk/voicemail/default/9373506524/unavail.slin' (language 'en')
> [Aug 23 15:50:32] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:4333 ast_rtp_read: 0x2addf402b830 -- Probation learning mode pass with source address 72.9.246.170:13730
> [Aug 23 15:50:37] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730
> [Aug 23 15:50:37] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730
> [Aug 23 15:50:38] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730
> [Aug 23 15:50:39] DEBUG[482]: chan_sip.c:4285 __sip_autodestruct: Auto destroying SIP dialog '0190242c0812028377b2281e2df47b3b@72.9.246.170:5060'
> [Aug 23 15:50:39] DEBUG[482]: chan_sip.c:6379 sip_pvt_dtor: Destroying SIP dialog 0190242c0812028377b2281e2df47b3b@72.9.246.170:5060
> [Aug 23 15:50:39] DEBUG[482]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x2addf4002d98'
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (58 requested / 58 actual) timer ticks per second
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:5414 set_format: Set channel SIP/voipms-00000046 to write format ulaw
> [Aug 23 15:50:39] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
>  
>    -- <SIP/voipms-00000046> Playing 'vm-intro.ulaw' (language 'en')
> [Aug 23 15:50:40] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:13730
> [Aug 23 15:50:40] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:13730
>  
> [Aug 23 15:50:42] DEBUG[482]: chan_sip.c:9057 find_call: = Looking for  Call ID: 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060 (Checking From) --From tag as2b5c0e97 --To-tag as7ac59689
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:28533 handle_incoming: **** Received BYE (8) - Command in SIP BYE
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '72.9.246.170:5060' into...
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '72.9.246.170' and port '5060'.
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:3387 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: res_rtp_asterisk.c:4755 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2addf4026628'
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:29442 stop_session_timer: Session timer stopped: 1 - 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:27149 handle_request_bye: Received bye, issuing owner hangup
> [Aug 23 15:50:42] DEBUG[482][C-00000052]: chan_sip.c:3731 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 72.9.246.170:5060
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: pbx.c:6789 __ast_pbx_run: Spawn extension (inbound,9373506524,5) exited non-zero on 'SIP/voipms-00000046'
>  == Spawn extension (inbound, 9373506524, 5) exited non-zero on 'SIP/voipms-00000046'
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:2662 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/voipms-00000046'
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: pbx.c:2111 new_find_extension: return at end of func
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:3595 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: channel.c:2841 ast_hangup: Hanging up channel 'SIP/voipms-00000046'
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: chan_sip.c:6929 sip_hangup: Hangup call SIP/voipms-00000046, SIP callid 7413649d1f4210266304a9fe5bfad539@72.9.246.170:5060
> [Aug 23 15:50:42] DEBUG[1723][C-00000052]: res_rtp_asterisk.c:4755 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2addf4026628'
> [Aug 23 15:50:42] DEBUG[438]: devicestate.c:345 _ast_device_state: No provider found, checking channel drivers for SIP - voipms
> [Aug 23 15:50:42] DEBUG[438]: chan_sip.c:29982 sip_devicestate: Checking device state for peer voipms
> [Aug 23 15:50:42] DEBUG[438]: devicestate.c:477 do_state_change: Changing state for SIP/voipms - state 1 (Not in use)
> [Aug 23 15:50:42] DEBUG[438]: devicestate.c:452 devstate_event: device 'SIP/voipms' state '1'
> [Aug 23 15:50:42] DEBUG[509]: app_queue.c:1924 handle_statechange: Device 'SIP/voipms' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
>  
> It doesn’t appear to be taking any action at all.  The system continues to record the message and delivers out to email.  Is it possible that the ‘a’ extension is broken?
>  
>  
> From: David Kerr [mailto:da...@ke...] 
> Sent: Wednesday, August 23, 2017 2:00 PM
> To: AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail server
>  
> That tells you that Asterisk is detecting the tone.  Doesn't tell you what it is doing with it... so you still need to trace dialplan execution (turn off debug, leave verbose on) to see what action it is taking on the tone.
>  
> David
>  
> On Wed, Aug 23, 2017 at 12:13 PM, Tim Turpin <tt...@z-...> wrote:
> I won’t copy in the entire session (way too much info), but here’s the result  of my pressing *,*,1,2,3,4,5,6,#.  It looks as though Asterisk is seeing the DTMF.
>  
>  
> [Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:12772
> 
> [Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:12772
> 
> [Aug 23 11:58:47] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at 72.9.246.170:12772
> 
> [Aug 23 11:58:48] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 42 (*), at 72.9.246.170:12772
> 
> [Aug 23 11:58:48] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 49 (1), at 72.9.246.170:12772
> 
> [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 49 (1), at 72.9.246.170:12772
> 
> [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 50 (2), at 72.9.246.170:12772
> 
> [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 50 (2), at 72.9.246.170:12772
> 
> [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 51 (3), at 72.9.246.170:12772
> 
> [Aug 23 11:58:49] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 51 (3), at 72.9.246.170:12772
> 
> [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 52 (4), at 72.9.246.170:12772
> 
> [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 52 (4), at 72.9.246.170:12772
> 
> [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 53 (5), at 72.9.246.170:12772
> 
> [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 53 (5), at 72.9.246.170:12772
> 
> [Aug 23 11:58:50] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 54 (6), at 72.9.246.170:12772
> 
> [Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 54 (6), at 72.9.246.170:12772
> 
> [Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating BEGIN DTMF Frame: 35 (#), at 72.9.246.170:12772
> 
> [Aug 23 11:58:51] DEBUG[1557][C-00000048]: res_rtp_asterisk.c:3591 create_dtmf_frame: Creating END DTMF Frame: 35 (#), at 72.9.246.170:12772
> 
>  
>  
>  
> From: David Kerr [mailto:da...@ke...] 
> Sent: Wednesday, August 23, 2017 11:05 AM
> To: AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] Question about setting up AstLinux as voicemail server
>  
> Check that the * key is not being captured for some other purpose (grep into other .conf files).  Check that you can match the * key outside of voicemail... use WaitExten() and validate that your dialplan sees that.  You can also go into the asterisk console ("asterisk -r") and turn on verbose and debug... e.g. "core set verbose 999" and "core set debug 999" and watch in the console.... make sure that logger.conf has a line that says "console => notice,warning,error,debug,verbose" else you might not get the debug and verbose messages into your console.
>  
> David
>  
> On Wed, Aug 23, 2017 at 10:53 AM, Tim Turpin <tt...@z-...> wrote:
> If I change my config to direct the call to VoiceMailMain(), I can log in
> with DTMF digits, so I know the carrier is passing tones. And Asterisk is
> recognizing them.
> Thanks.
> 
> -----Original Message-----
> From: Lonnie Abelbeck [mailto:li...@lo...]
> Sent: Wednesday, August 23, 2017 10:51 AM
> To: AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] Question about setting up AstLinux as
> voicemail server
> 
> Tim,
> 
> Make sure in your sip.conf for your inbound provider the setting for
> "dtmfmode" matches what your provider requires, Asterisk defaults to rfc2833
> .
> 
> Lonnie
> 
> 
> On Aug 23, 2017, at 9:20 AM, Tim Turpin <tt...@z-...> wrote:
> 
> > Getting closer, I think.
> >
> > I'm starting to wonder if the DTMF '*' is being recognized at all.  Now
> the caller is dropped into the proper mailbox, but pressing '*' does
> nothing.
> > Here's extensions.conf:
> >
> > [inbound]
> >
> > exten => _NXXNXXXXXX,1,Answer
> > exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call)
> > exten => _NXXNXXXXXX,n,Set(boxnumber=${EXTEN})    ; set a variable for box
> number
> > exten => _NXXNXXXXXX,n,NoOp(${boxnumber})        ;  test for variable
> > exten => _NXXNXXXXXX,n,Voicemail(${boxnumber})
> > exten => _NXXNXXXXXX,n,Hangup
> > exten => a,1,VoiceMailMain(${boxnumber})      ; user dialed * in
> greeting. send them to their mailbox
> > exten => a,n,Hangup
> >
> >
> > Here's the response when calling the DID number 9373506524:
> >
> > Connected to Asterisk 11.25.1 currently running on SST (pid = 415)
> >  == Using SIP RTP CoS mark 5
> >    -- Executing [9373506524@inbound:1] Answer("SIP/voipms-00000037", "")
> in new stack
> >    -- Executing [9373506524@inbound:2] NoOp("SIP/voipms-00000037",
> "inbound-phone-call") in new stack
> >    -- Executing [9373506524@inbound:3] Set("SIP/voipms-00000037",
> "boxnumber=9373506524") in new stack
> >    -- Executing [9373506524@inbound:4] NoOp("SIP/voipms-00000037",
> "9373506524") in new stack
> >    -- Executing [9373506524@inbound:5] VoiceMail("SIP/voipms-00000037",
> "9373506524") in new stack
> >    -- <SIP/voipms-00000037> Playing
> '/var/spool/asterisk/voicemail/default/9373506524/temp.slin' (language 'en')
> >    -- <SIP/voipms-00000037> Playing 'vm-intro.ulaw' (language 'en')
> >    -- <SIP/voipms-00000037> Playing 'beep.ulaw' (language 'en')
> >    -- Recording the message
> >    -- x=0, open writing:
> > /var/spool/asterisk/voicemail/default/9373506524/tmp/2u8Hzw format:
> > wav, 0x2addfc001798
> >
> > Is there any setting that would not allow the '*' to be recognized during
> the greeting?
> >
> >
> >
> >
> > From: The Cadillac Kid via Astlinux-users
> > [mailto:ast...@li...]
> > Sent: Wednesday, August 23, 2017 8:32 AM
> > To: AstLinux Users Mailing List
> > Cc: The Cadillac Kid
> > Subject: Re: [Astlinux-users] Question about setting up AstLinux as
> > voicemail server
> >
> > set a variable first... the issue is that ${EXTEN} changes to 'a' when you
> * out...  ${EXTEN} is the current extension you are workign with and you
> want to go to the original dialed extension.
> >
> > [inbound]
> > exten => _NXXNXXXXXX,1,Answer
> > exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call)
> > ; set a variable for box number
> > exten => _NXXNXXXXXX,n,Set(boxumber=${EXTEN})
> >
> > exten => _NXXNXXXXXX,n,Voicemail(${boxnumber})
> > ;exten => _NXXNXXXXXX,n,VoiceMailMain(${EXTEN})
> > exten = > _NXXNXXXXXX,n,Hangup
> >
> > ; user dialed * in greeting. send them to their mailbox
> >
> > exten => a, 1, VoicemailMain(${boxnumber}) exten => a,n, Hangup
> >
> >
> >
> > -Christopher
> >
> >
> > From: Tim Turpin <tt...@z-...>
> > To: 'AstLinux Users Mailing List'
> > <ast...@li...>
> > Sent: Wednesday, August 23, 2017 8:14 AM
> > Subject: Re: [Astlinux-users] Question about setting up AstLinux as
> > voicemail server
> >
> > This appears to possibly work for one mailbox user.  We have a couple
> thousand users, all dialing in via DID, and the process needs to be the same
> for all users.  My current extensions.conf looks like this:
> >
> > [inbound]
> > exten => _NXXNXXXXXX,1,Answer
> > exten => _NXXNXXXXXX,n,NoOp(inbound-phone-call)
> > exten => _NXXNXXXXXX,n,Voicemail(${EXTEN}) ;exten =>
> > _NXXNXXXXXX,n,VoiceMailMain(${EXTEN})
> > ;exten => a, 1, VoicemailMain(${EXTEN})
> >
> > I've played with the 'a' extension in different formats, but can't seem to
> make it work.  In the current configuration, when a caller dials in, it
> plays the greeting for that particular mailbox.  If I comment out the third
> line and un-comment the fourth, the caller drops into their box with the
> ability to log in.  I can't figure out how to utilize the 'a' extension to
> allow the user to press '*' to login while listening to his greeting (the
> fifth line).
> >
> > I'm using information about the 'a' extension from the following sites:
> >
> > From ' https://www.voip-info.org/wiki-asterisk+standard+extensions1 ':
> > a: Called when user presses '*' during a voicemail greeting
> > h: Hangup extension
> > i: invalid extension
> > o: Operator extension, used for operator exit by pressing zero in
> > voicemail
> > s: Start extension in context
> > t: Timeout extension
> > T: AbsoluteTimeout() extension
> > Also, from ' https://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail ':
> > Also. during the prompt if the caller presses:
> > '*' - the call jumps to extension 'a' in the current voicemail context.
> > Example:
> > Exten => a, 1, VoicemailMain(@default) Exten => a, 2, Hangup Being a
> > novice at Asterisk, I have to assume that I'm not following the proper
> coding format, or I'm not applying the 'a' extension properly.  From what I
> have read on these two web pages, I think that this is the application to
> use, but I'm just not applying it properly.
> >
> > From: David Kerr [mailto:da...@ke...]
> > Sent: Tuesday, August 22, 2017 5:30 PM
> > To: AstLinux Users Mailing List
> > Subject: Re: [Astlinux-users] Question about setting up AstLinux as
> > voicemail server
> >
> > Tim,
> >  You are going to want to use the Background() app to play the greeting
> with the WaitExten() app to wait for a keypress (if they wait til the very
> end of the greeting before pressing) and then the Authenticate() app to get
> a PIN to proceed to whatever action is permitted.  Something like this
> (untested but should be close enough)...
> >
> > [leavemessage]
> > exten = s,1,NoOp(voicemail)
> >  same = n,Ringing()
> >  same = n,Wait(2)
> >  same = n,Answer()
> >  same = n(start),Set(TIMEOUT(response)=1)  same =
> > n,Set(TIMEOUT(digit)=1)  same = n,Background(record/NoAnswer) ; my
> > custom message, press 1 or wait to leave a msg  same = n,WaitExten(1)
> > exten = 1,1,Voicemail(101,us) ; caller pressed 1  same = n,NoOp(Back
> > from voicemail)  same = n,Hangup() exten =
> > _[*],1,VoiceMailMain(101,sa(0)) ; caller pressed *  same = n,NoOp(Back
> > from voicemailmain)  same = n,Hangup() exten = t,1,Voicemail(101,us) ;
> > timeout, leave a message. could GoTo(1,1)  same = n,NoOp(Back from
> > voicemail)  same = n,Hangup() exten = i,1,Playback(pbx-invalid) ;
> > standard invalid key pressed msg.
> >  same = n,Goto(s,start)
> > exten = h,1,Hangup()
> >
> > David
> >
> >
> >
> > On Tue, Aug 22, 2017 at 3:04 PM, Tim Turpin <tt...@z-...> wrote:
> > Thank you for the fast reply.
> >
> > I loaded up the AstLinux last week. I've been able to figure out most
> > of what I need, except for a way to route incoming DID calls to
> > voicemail, allowing the caller to be able to press '*' while hearing
> > the mailbox greeting and then be handed off to 'VoiceMailMain()' to log
> into their box.
> > If '*' isn't pressed, the caller would just drop into the mailbox to
> > leave a message.
> >
> > It seems like it should be easy to set up, but it's really kicking my
> > butt right now, and I'm just trying to determine my best avenue for
> > assistance in figuring this out.  I'll try the Asterisk forums and see
> > if they can offer any help.
> >
> > Thanks again.
> >
> > Tim
> >
> >
> >
> > -----Original Message-----
> > From: Lonnie Abelbeck [mailto:li...@lo...]
> > Sent: Tuesday, August 22, 2017 2:31 PM
> > To: AstLinux Users Mailing List
> > Subject: Re: [Astlinux-users] Question about setting up AstLinux as
> > voicemail server
> >
> >
> > On Aug 22, 2017, at 11:49 AM, Tim Turpin <tt...@z-...> wrote:
> >
> > > I'm new to the Asterisk world, and I'm trying to use AstLinux to
> > > replicate
> > an existing voicemail environment, and I have several configuration
> > questions.  Is this the proper forum for these questions, or do I send
> > the questions somewhere else?
> > >
> > > Thanks.
> > > Tim.
> >
> > Hi Tim,
> >
> > First, using AstLinux as a dedicated voicemail server, using a small
> > x86 appliance and SSD storage or Virtual Machine Guest is a good approach.
> >
> > This mailing list is mostly dedicated to AstLinux Project specific
> > questions, Asterisk voicemail.conf, sip.conf and extensions.conf
> > configurations are best asked in the Asterisk support groups.  If you
> > have things all but working and have reached a brick wall using
> > AstLinux ... you can give this list a try.
> >
> > Keep in mind that using AstLinux, you will be required to generate the
> > base extensions.conf text file for yourself, AstLinux has a basic web
> > interface and "Users" tab that can help manage your voicemail users.
> > As a starting point you might spin-up the "Guest VM x86-64bit (Video
> > Console)" Install ISO in a virtual machine to give you a playground to
> > test before purchasing any hardware.
> >
> > Alternatively, if coding a extensions.conf is not your cup-of-tea you
> > might query this mailing list for off-line consulting help.
> >
> > Here is a reference to give you the flavor of the configuration ...
> >
> > Configuring Voice Mail Boxes
> > https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxe
> > s
> >
> > Lonnie
> >


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[Astlinux-users] Fwd: Can't log into user interface astlinux

[Roberto R. <rri...@gm...>]

Alix 6f2 = 2 LAN / 1 miniPCI / 1 
miniPCI Express / LX800 / 256 MB / USB / dual SIM socket ....this is what I'm using for my pbx.

Thanks all

Sent from my iPhone

Begin forwarded message:

> From: Roberto Rivera <rri...@gm...>
> Date: August 23, 2017 at 4:43:31 PM EDT
> To: AstLinux Users Mailing List <ast...@li...>
> Subject: Can't log into user interface astlinux
> 
>> Hi all,
>> I was using astlinux then got a msg that I had no memory so it stopped everything. I rebooted and everything was fine....it happened again but now I can't get access to the site for the past day it so....I can make outbound calls and was getting ready to work on getting inbound working when I noticed the problem any suggestion would be greatly appreciated...ip address has not changed and I can't ssh in either.
>> Thanks
> 
> Sent from my iPhone