hello All
I am using festival as an application
but it default voice is not good to hear
anybody have solution about better voice in Festival
regards
Dhaval
___
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asterisk-users
Your ITSP is giving you the DNIS digits. You have to match them in your
dialplan.
What if your ITSP routed calls from FIFTY different numbers to your switch?
How would you differetntiate between them if they all just routed to the S
extension? That's why the paradigm is based on passing and
Hello,
I wanted to add a failover trunk to my asterisk configuration.
I got 2 gateways for my calls.. one is a pri other is voip trunk.
I want to keep my trunk for failover.
I am using ast 1.6 with asterisk-gui.
But when i add a failover trunk for test purposes asterisk-gui adds the
following line
On Wed, Jun 17, 2009 at 03:35:56PM -0700, bilal ghayyad wrote:
Hi All;
asterisk-gui read/write from the conf files or database?
Asterisk-gui's database is the config files ;-)
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
2009/6/18 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Wednesday 17 June 2009 17:06:25 Olivier wrote:
At the moment, I can't read Local channels variables using IMPORT
function
: ${IMPORT(Local/7...@pcdialer-5dff\;1,CALLERID(num))}
I'm confused as to why you're trying to escape
On Wed, Jun 17, 2009 at 04:09:46PM -0700, Darryl Dunkin wrote:
hardhdlc is for a BRI, use dchan=24 instead to set the d-channel.
hardhdlc is *NOT* BRI-specific.
In ISDN (BRI/PRI) messages on the D channel are encoded in HDLC:
http://en.wikipedia.org/wiki/HDLC
Zaptel originally decoded the
What happens if the http server is down? My point is that I don't
want it
to try and pull any config from a server. I just want it to use
its local
config.
I don't recall this looping probelm. The value of tries is supposed to
prevent this from happening.
r
On Wed, Jun 17, 2009 at 01:08:33PM -0500, Danny Nicholas wrote:
If this is a recorded sound, you might want to truncate it with lame or
audacity. It is quite common in my shop as we record using the phones.
Thanks for this suggestion.
The problem was indeed a silence at the beginning of my
Hi all,
I'm trying to connect one phone to a remote asterisk server via openvpn.
First of all, I put the vpn server on the box hosting asterisk and the
vpn client on another box, both with public ips.
Then I set the client ip as my phone IP gateway and the remote pbx ip as
the registrar and
I am using FreePBX with Asterisk 1.4 and i wanted to upgrade to Asterisk
1.6.1.
As FreePBX only supports ZAP naming i set dahdichanname = no in my
asterisk.conf.
However, after installation the console was still merrily chattering about
incoming calls on DAHDI channels and nothing happened
Usually this is a routing error with openvpn setup and asterisk thinking
it needs to route someway other than the vpn. If the originating packets
have an external ip address asterisk might send them back out another route
Have a look using tcpdump on the server to see where the returned
Greetings everyone,
I'm trying to compile asterisk for an AVR32 (Atmel NGW100).
Buildroot for AVR32 already has the asterisk package, though it has
bugs. Firstly it tries to apply a patch for 1.2 on a 1.6, but deleting
the contents of the patch file did the trick.
Now, the problem is making
Dear all,
I am currently trying to configure a PBX make use of a multiple of
outgoing lines, currently my extensions.conf looks something like below
; extensions.conf
; 20th October 2008
[globals]
sip1=201
sip2=202
sip3=203
sip4=204
[general]
autofallthrough=yes
Hi Clara,
You could put some data into astdb and query for the outgoing line and
callerid based on internal callerid (extension).
something like
user/201/outline 89859715
user/201/outcallerid 89859715
and so on...
By the way: _89859715 without the dot (.) is same like 89859715 - maybe
you
On Thu, 2009-06-18 at 03:50 +, Joseph L. Casale wrote:
I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com.
The Asterisk console shows:
[Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '36' rejected because extension
Hi
I am trying to setup asterisk to do a mass deploy of some snom phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where to set the notify reply.
I was hoping to not have to use dhcp options
alex
signature.asc
Description: Digital signature
I thought TFTP (and therefore, DHCP option 66) is the only
autoprovisioning method Asterisk supports?
--
Sent from mobile device
On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote:
Hi
I am trying to setup asterisk to do a mass deploy of some snom
phones. I
can't find where
Remco Barendse wrote:
I am using FreePBX with Asterisk 1.4 and i wanted to upgrade to Asterisk
1.6.1.
As FreePBX only supports ZAP naming i set dahdichanname = no in my
asterisk.conf.
However, after installation the console was still merrily chattering about
incoming calls on DAHDI
On Thu, 2009-06-18 at 10:31 +0200, Giorgio Incantalupo wrote:
Hi all,
I'm trying to connect one phone to a remote asterisk server via openvpn.
First of all, I put the vpn server on the box hosting asterisk and the
vpn client on another box, both with public ips.
Then I set the client ip
hello,
you can define a variable in sip.conf in each extension like:
[201]
...
setvar=LINE=89859716
...
then in extensions when user 201 calls you have a the var defined and you can
use it with ${LINE}.
On Thu, Jun 18, 2009 at 08:19:27PM +1000, Clara Chan wrote:
Dear all,
I am
On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote:
I am trying to setup asterisk to do a mass deploy of some snom
phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where to set the notify reply.
I was hoping to not have to use dhcp
Hi John,
I already have the ccd dir with the iroute (mandatory for routing to
pc/phone connected to vpn client). During the last test I could register
and make a call but voice disappears after 1, 2 seconds. I'm trying to
understand if it is a bandwidth problem. At the moment I have my phone
On Thu, 18 Jun 2009, Kevin P. Fleming wrote:
Remco Barendse wrote:
I am using FreePBX with Asterisk 1.4 and i wanted to upgrade to Asterisk
1.6.1.
As FreePBX only supports ZAP naming i set dahdichanname = no in my
asterisk.conf.
However, after installation the console was still merrily
Have you tried #1103 or *2103? The # would do a blind transfer, the * would
initiate an attended transfer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad
Sent: Wednesday, June 17, 2009 9:41 PM
To:
If you like the voice, but it is just too low, you can amplify the Festival
output with sox (sox -V 3 softer.wav louder.wav)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, June 18, 2009
Do you have 'canreinvite=no' in your sip.conf entry for this phone? If
not, you should.
On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote:
Hi John,
I already have the ccd dir with the iroute (mandatory for routing to
pc/phone connected to vpn client). During the last test I could register
It depends on how you are configured. The gui interfaces using Asterisk
Manager, so you get the Same IO from the gui that you would get from a
native manager session.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
--Original Message Text---
From: Doken, Serhad
Date: Wed, 17 Jun 2009 16:07:12 -0700
Hi,
I wanted to follow up on this thread about WB support on the MeetMe
bridge that is in 1.6.2. Does it only work for G722 or any WB codec ?
I am working with another 16k WB codec that I can transcode to
Hi,
I really need some help, I can't find the way to install Noojeefax.
I have the files from sourceforge but there is no readme to explain what
to do, and no help on the web...
thanks
Will
!DSPAM:4a3a3e6763731933410313!
___
-- Bandwidth and
Have you tried this page? http://www.asteriskit.com.au/Page/AsterFax
Why don't you just freefaxforasterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of wilfried
bordoni
Sent: Thursday, June 18, 2009 8:17
Yes I already saw every page of the web containing noojeefax ...
I need a mail to fax, and I think Noojeefax is the only one to provide that.
!DSPAM:4a3a434263739759639719!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hi Darrick,
I always set canreinvite=no 'cause it gives a lot of problems if set to
yes (and the default is).
I made a call with rtp debug on and I noticed that normally, on the
asterisk CLI, I see one packet sent corresponding to one packet got
(made a test with a local call on our
Giorgio,
tcpdump and wireshark are your friends. Instead of guessing, capture a
call with tcpdump then look at it with wireshark.
Darrick
On 06/18/2009 08:58 AM, Giorgio Incantalupo wrote:
Hi Darrick,
I always set canreinvite=no 'cause it gives a lot of problems if set to
yes (and the
We are trying to configure Asterisk (version 1.6.1.0) with some SIP
phones behind a SIP Proxy/NAT device. The phones register properly to
Asterisk, and to get Asterisk to register properly to the external SIP
registrar we added this to the general section of sip.conf (the address
of the
There are some things that are not that clear to me :
When I want to write CDR-info to an external MySQL-DB
- do I need to install the asterisk-addons prior to installing Asterisk
or after having installed Asterisk ??
- How do I tell Asterisk not to write CDR-info to the Master.csv file
but into
John A. Sullivan III wrote:
Hello, all. I am delightfully slogging my way through installing and
configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and
admiring the product but I'm having a problem getting speex to install
and I would very much like to use it. It is not available
Hello, all. I am delightfully slogging my way through installing and
configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and
admiring the product but I'm having a problem getting speex to install
and I would very much like to use it. It is not available in menuselect
and the problem
On Thursday 18 June 2009 10:08:44 jonas kellens wrote:
There are some things that are not that clear to me :
When I want to write CDR-info to an external MySQL-DB
- do I need to install the asterisk-addons prior to installing Asterisk
or after having installed Asterisk ??
After. Addons
On Thu, Jun 18, 2009 at 12:41:39PM -0400, John A. Sullivan III wrote:
[r...@pbx01 ~]# rpm -qa | grep speex
speex-devel-1.0.5-4.el5_1.1
speex-1.0.5-4.el5_1.1
That is too old a version. speex 1.1.x will happen to work. 1.0.x will
not have the newer DSP interface. It does have the basic Speex
On Thu, 2009-06-18 at 19:58 +0300, Tzafrir Cohen wrote:
On Thu, Jun 18, 2009 at 12:41:39PM -0400, John A. Sullivan III wrote:
[r...@pbx01 ~]# rpm -qa | grep speex
speex-devel-1.0.5-4.el5_1.1
speex-1.0.5-4.el5_1.1
That is too old a version. speex 1.1.x will happen to work. 1.0.x will
On Thu, Jun 18, 2009 at 12:46 PM, Brent Davidson
br...@texascountrytitle.com wrote:
John A. Sullivan III wrote:
Hello, all. I am delightfully slogging my way through installing and
configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and
admiring the product but I'm having a
Hi there,
we have a problem with dahdi and overlapdial. We are running an E1 in
Germany and are in need of overlapdial. The E1 is connected to a Sangoma
A101.
As soon as overlapdial is set to yes we have problems with incoming
audio on the dahdi channels. When set to no all audio is fine.
Does anyone know of a way to force the voicemail password for users to be of a
certain length? We've setup operator=yes within our voicemail.conf and want to
have the users use a long password to prevent possible guessing by external
parties. I'm not seeing any such option in my research. If it
AFAIK, this doesn't exist. However, you could disable password changing in
the voicemail application and set it from the dialplan and force a minimum
length there.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin
Forgot to add:
Asterisk full log only shows no anomalies. Normal call clearing when you
hangup, nothing else.
Regards
Bjoern Metzdorf wrote:
Hi there,
we have a problem with dahdi and overlapdial. We are running an E1 in
Germany and are in need of overlapdial. The E1 is connected to a
!! Got reject for frame 61, but we have nothing -- resetting!
!! Got reject for frame 63, but we have nothing -- resetting!
!! Got reject for frame 65, but we have nothing -- resetting!
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
Connect Request
q931.c:3009 q931_disconnect:
Hi
I am trying to implement monitoring of asterisk (all 4 spans-i want to show
them line by line Up or down) using nagios using below script, but i always get
the status as down and red..can anyone let me know how to read an output from
nagios plugin ? nagios etc is configured already and is
On Thu, 2009-06-18 at 13:17 -0400, Steve Totaro wrote:
On Thu, Jun 18, 2009 at 12:46 PM, Brent Davidson
br...@texascountrytitle.com wrote:
John A. Sullivan III wrote:
Hello, all. I am delightfully slogging my way through installing
and
configuring
I didn't have a limit set, but I put one on of 5 for testing sake that
didn't change a thing.
James Shigley
Monroe Telephone Answering Service
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, June
Check the script permissions for nagios user
Sriram escreveu:
Hi
I am trying to implement monitoring of asterisk (all 4 spans-i want to
show them line by line Up or down) using nagios using below script, but
i always get the status as down and red..can anyone let me know how to
read an
On Thu, 18 Jun 2009, Darrin Henshaw wrote:
Does anyone know of a way to force the voicemail password for users to
be of a certain length? We've setup operator=yes within our
voicemail.conf and want to have the users use a long password to prevent
possible guessing by external parties. I'm
On Wed, Jun 17, 2009 at 7:41 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Wed, Jun 17, 2009 at 3:18 PM, John Todd jt...@digium.com wrote:
On Jun 17, 2009, at 8:16 AM, Steve Totaro wrote:
Hi,
Quick question to the real world.
Approx what specs would I need on server
I am calling CHARLOT,DANIEL @ 4099819921 to tell him that you posted his
name and phone number on the interweb
Sometimes a redaction is prudent.
On Thu, Jun 18, 2009 at 2:24 PM, James A. Shigley j...@answeringserv.comwrote:
!! Got reject for frame 61, but we have nothing -- resetting!
!!
Hi,
after further investigation we found a solution:
overlapdial=incoming
See also https://issues.asterisk.org/view.php?id=7511
Regards,
Bjoern
Bjoern Metzdorf wrote:
Forgot to add:
Asterisk full log only shows no anomalies. Normal call clearing when you
hangup, nothing else.
Regards
Why not connect to the AMI via telnet?
On Thu, Jun 18, 2009 at 2:27 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi Danny;
Really I did not understand how I can determine if the IO will be DB or
conf files? Is it from the Asterisk manager?
Regards
Bilal
-
It
Hi Danny;
Really I did not understand how I can determine if the IO will be DB or conf
files? Is it from the Asterisk manager?
Regards
Bilal
-
It depends on how you are configured. The gui interfaces using Asterisk
Manager, so you get the Same IO from the gui that
I don't feel like looking it up but does a capital G and lowercase g in your
DAHDI/group make a difference?
Just a thought.
On Thu, Jun 18, 2009 at 2:28 PM, James A. Shigley j...@answeringserv.comwrote:
I didn’t have a limit set, but I put one on of 5 for testing sake that
didn’t change a
G looks up 1,2,3,4,5, g looks up 5,4,3,2,1 so yes, at least in theory. If
you only have one open line, no harm no foul.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Thursday, June 18, 2009 1:28 PM
To:
It errors the same whether I use g or G.
James Shigley
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Totaro
Sent: Thursday, June 18, 2009 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
As usual my manager comes up with some obscure reference I didn't find. There
seems to be a parameter called minpassword described here:
http://www.asterisk.org/doxygen/trunk/Config_vm.html
But from further digging it looks like it's a 1.6.1.0 feature. Might see about
a backport if possible.
On Thu, 18 Jun 2009, Sriram wrote:
I am trying to implement monitoring of asterisk (all 4 spans-i want to
show them line by line Up or down) using nagios using below script, but
i always get the status as down and red..can anyone let me know how to
read an output from nagios plugin ?
On Thu, 2009-06-18 at 14:21 +0200, Philipp Kempgen wrote:
On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote:
I am trying to setup asterisk to do a mass deploy of some snom
phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where to set
On Wednesday 17 June 2009 16:54:53 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 15:43 -0500, Tilghman Lesher wrote:
On Wednesday 17 June 2009 14:55:55 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 14:18 -0500, Tilghman Lesher wrote:
On Wednesday 17 June 2009 11:56:28 John A.
On Thu, 2009-06-18 at 14:20 -0500, Tilghman Lesher wrote:
On Wednesday 17 June 2009 16:54:53 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 15:43 -0500, Tilghman Lesher wrote:
On Wednesday 17 June 2009 14:55:55 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 14:18 -0500, Tilghman
I am not able to understand the relation between the AMI and the GUI? And
really I am not able to know where to determine if my Asterisk will read/write
with DB or with config files?
Regards
Bilal
---
Why not connect to the AMI via telnet?
On Thu, Jun 18,
Here's the .05 tour as I know it:
Every Asterisk installation reads /etc/asterisk/asterisk.conf and
/etc/extconfig.conf at startup. Once these files are parsed, the remaining
IO is done based off of how these two files point.
When you run the GUI, it does a series of AMI calls to get it's
Thanks Michael. I guess prior to 1.6.2, Asterisk was downgrading streams to
SLIN before mixing and then mixed stream got upgraded to WB.
My question is, with this release, is Asterisk converting WB codecs to SLIN16
and mix them that way ? That seems to be the logical way to me just wanted an
Loan,
Thanks for your help in this matter.
Having never used astdb before, can you point me to an example on this??
Thanks hugely,
Clara
Hi Clara,
You could put some data into astdb and query for the outgoing line and
callerid based on internal callerid (extension).
something like
On Thu, Jun 18, 2009 at 11:27:18AM -0700, bilal ghayyad wrote:
Hi Danny;
Really I did not understand how I can determine if the IO will be DB or
conf files? Is it from the Asterisk manager?
Again, unless you make some pretty major changes in the way the
asterisk-gui[1] works, it will use
Protocol Discriminator: Q.931 (8) len=5
Call Ref: len= 2 (reference 27418/0x6B1A) (Originator)
Message type: CONNECT ACKNOWLEDGE (15)
!! Got reject for frame 69, but we have nothing -- resetting!
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
Connect Request
The remote
On Thu, Jun 18, 2009 at 07:34:38AM -0400, Alex Balashov wrote:
I thought TFTP (and therefore, DHCP option 66) is the only
autoprovisioning method Asterisk supports?
seems like the documentation from snom for V7, includes the pnp method
as well. it sends a subscribe to a multicast address
Conrad Wood schrieb:
On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote:
I am trying to setup asterisk to do a mass deploy of some snom
phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where to set the notify reply.
FWIW I use a
Hello List;
Actually based on what I read at Guru that after I did the installation and
configuration of the asterisk-gui, I can access it using the link:
http://id_address:8088/asterisk/static/config/cfgadvanced.html
I tried to search for something like
/var/lib/asterisk/static-http/config/cfgadvanced.html is the file location.
The root directory is /var/lib/asterisk unless you change it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent:
Alex Samad schrieb:
seems like the documentation from snom for V7, includes the pnp method
as well. it sends a subscribe to a multicast address (224.0.1.75) and the
listener is
meant to respond with a notify which has the url which is normally sent
my dhcp, i was hoping to use that
I
On Jun 18, 2009, at 2:57 PM, Philipp Kempgen wrote:
I think I would prefer this method, but I can't find where to set
asterisk to listen to the multicast address nor where to program the
notify reply
I have already told you that Asterisk is not involved in the process
of configuring the
Clara Chan wrote:
Loan,
Thanks for your help in this matter.
Having never used astdb before, can you point me to an example on this??
Thanks hugely,
Clara
Clara --
You need to read the book. In it you'll find examples.
Asterisk: The Future of Telephony 2nd Edition (ISBN
I'm told that Asterisks wideband capability is exclusively based upon
16 khz sampling. Higher sampling rates, like you might find with CELT,
are downsampled for mixing at 16 khz.
My guess is that not everyone will be happy with this, especially
vendors trumpeting codecs with higher sampling
On Thu, 2009-06-18 at 14:55 +0200, Giorgio Incantalupo wrote:
Hi John,
I already have the ccd dir with the iroute (mandatory for routing to
pc/phone connected to vpn client). During the last test I could register
and make a call but voice disappears after 1, 2 seconds. I'm trying to
I believe that 'externpasscheck' was added in the 1.6 branch. Since we use
this, I wrote a quick perl script that checks for password length,
difficulty, repeated digits, etc. which are required for us. If you get it
back-ported to the version you are on you can have the script, just contact
me
On Thu, Jun 18, 2009 at 08:06:24AM -0500, Danny Nicholas wrote:
Have you tried #1103 or *2103? The # would do a blind transfer, the * would
initiate an attended transfer.
I tried combination of
flash extention
flash *67 exten
extention
*67 exten
and re iterated with # instead of *.
The
On Thu, Jun 18, 2009 at 11:57:20PM +0200, Philipp Kempgen wrote:
Alex Samad schrieb:
seems like the documentation from snom for V7, includes the pnp method
as well. it sends a subscribe to a multicast address (224.0.1.75) and the
listener is
meant to respond with a notify which has the
On Thu, Jun 18, 2009 at 02:21:47PM +0200, Philipp Kempgen wrote:
On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote:
I am trying to setup asterisk to do a mass deploy of some snom
phones. I
can't find where i configure asteriks to listen to the multicast
address, nor where
Hi
I have 2 digium cards (tdm410) with combination of fxs + multiple fxo
ports.
I have had a quick look at sangoma B series cards. I was wondering if
there is a card out there with
hardware echo canceller
say max 4 ports (mix of fxs/fxo)
g729 encoding onboard
Alex
--
More and more of our
After a kernel update (but before rebooting) Is there a way to recompile
Zap/Dahdi against the new kernel?
My objective is to eliminate the additional downtime that occurs while
recompiling/installing zap/dahdi after booting into the new kernel.
Please correct me if I'm wrong:
My
Hi All,
How can I force an anonymous SIP connection from a certain IP address to use a
specific context rather than the default one defined in sip.conf.
I am using Asterisk 1.6.0.9
Regards
David Klaverstyn
___
-- Bandwidth and Colocation Provided
On Thu, Jun 18, 2009 at 11:24:40PM -0500, Karl Fife wrote:
After a kernel update (but before rebooting) Is there a way to recompile
Zap/Dahdi against the new kernel?
My objective is to eliminate the additional downtime that occurs while
recompiling/installing zap/dahdi after booting into
Hello
I found following issue while trying to load flite module from CLI
module load app_flite.so
Unable to load module app_flite.so
Command 'module load app_flite.so ' failed.
[Jun 19 11:31:12] WARNING[23507]: loader.c:384 load_dynamic_module: Module
'app_flite.so' did not register itself
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