+01:00 Ron Wheeler <rwhee...@artifact-software.com
<mailto:rwhee...@artifact-software.com>>:
CentOS 7 works well with Asterisk.
Install latest CentOS7 with updates install asterisk
I am running FreePBX on CentOS 7.
Ron
On 14/12/2017 10:38 AM, Olivier wrote
3 255 7799
Mobile +972 54 666 0933
_Email _e...@greenfield.tech <mailto:e...@greenfield.tech>
Skype: EricLKlein
Web: www.greenfield.tech <https://www.greenfield.tech/>
www.cloudonix.io <http://www.cloudonix.io/>
Message: 1
Date: Fri, 15 Dec 2017 09:42:00 -0500
traffic crosses many networks.
Am I way off track?
Ron
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7 build 1701. After the install, I apply
updates as they are issues by the CentOS team.
Works fine.
Ron
On 15/12/2017 5:59 AM, Olivier wrote:
Hello Ron,
Which kernel do you run Asterisk/Freepbx with ?
Cheers
2017-12-14 16:57 GMT+01:00 Ron Wheeler <rwhee...@artifact-software.
.
After a short research (see [1] as an example), I'm wondering what are
general kernel practices on CentOS regarding Asterisk and when
targeting stability:
- Is it recommended to upgrade kernel version(s) (ie moving from linux
3.10 to 4.3) just after OS installation ?
Best regards
--
Ron
to Ron's phone and
yyy-y to go to Paul?
Both should be easy to do and not too hard to set up under FreePBX.
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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
be able to help you.
Surely, you mean the Biz List
Doug
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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
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wrote:
I know but this is not my sole decision.
On 12.12.2017 16:17, Ron Wheeler wrote:
If your phone system goes down and you can not get it back up until
tomorrow afternoon because your support person is on another project,
you may wish you had an SLA.
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Ron Wheeler
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Artifact
Phones ( 1 dect)
- Create new voip trunk (current are ISDN) (30 number block)
- LTS is important
- an SLA is optional at the moment there is no one
On 11.12.2017 22:31, Ron Wheeler wrote:
You might want to add some details
- size of the project
-- number of locations
-- number of extensions
for Support and a
Phonehardware Reseller in Germany.
The should be no license costs per User / Server.
- Install Configure Asterisk for our specification
- Install FreePBX or similar (optional)
- Resell Hardware
Thanks for any suggest.
Best Regards,
basti
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*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ron
Wheeler
*Sent:* Friday, December 08, 2017 03:06 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Showing CallerID on multiple phones
I could be way off track but have you looked a
with different CRM packages or Outlook,
but I was wondering if a way was available to show the calls on their
phones.
Thanks in Advance;
John V.
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(journalctl).
If your question has changed (you mentioned ‘the first problem’) then
ignore the above; jumping in late.
**Jason**
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skype: ronaldmwheeler
phon
post the relevant portion of your syslog
(journalctl).
If your question has changed (you mentioned ‘the first problem’) then
ignore the above; jumping in late.
**Jason**
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phone: 866-970
t(1489588785.834:1183): arch=c03e syscall=2 success=no exit=-13 a0=1be0de0
a1=8 a2=1a4 a3=1be0de0 items=0 ppid=1485 pid=3857 auid=4294967295 uid=0 gid=0 euid=0 suid=0
fsuid=0 egid=0 sgid=0 fsgid=0 tty=(none) ses=4294967295 comm="asterisk"
exe="/usr/sbin/asterisk" subj=sys
/audit/audit.log
and try the command.
Is there any open bug for a security policy for Asterisk?
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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
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the command.
Is there any open bug for a security policy for Asterisk?
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President
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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
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NING[2332]: chan_sip.c:4061 retrans_pkt:
Retransmission timeout reached on transmission
56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request)
-- See >>> >>>
Firewall?
Doug
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Ron Wheeler
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email: rwhee...@artifact-software.com
and they would
register to the Public Asterisk server from that IP address.
Should I move the asterisk server behind the same router?
If so, how should the server be set up and what is the best
router/firewall hardware to accomplish this environment?
Thanks,
-H
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Ron Wheeler
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have Centos 6 with MySQL 5.1.73 and PHP 5.3.3 with FreePBX 2.11.
Ron
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President
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email: rwhee...@artifact-software.com
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On 27/07/2015 2:38 PM, Steve Edwards wrote:
On 27/07/2015 1:51 PM, Steve Edwards wrote:
Any particular reason CentOS 7 repos aren't available?
I'm finding integration issues with CentOS 6's ancient versions of
MySQL and PHP with third party applications.
On Mon, 27 Jul 2015, Ron Wheeler
*From*: Ron Wheeler rwhee...@artifact-software.com
*Sent*: Thursday, March 12, 2015 9:40 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] switching from SIP to Skype..or not
Your characterization may be true
Bryant
*From*: Ron Wheeler rwhee...@artifact-software.com
*Sent*: Thursday, March 12, 2015 9:40 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] switching from SIP to Skype..or not
Your characterization
what it was designed for.
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servers ;-) ?
Thanks!
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Moshiach is coming very soon, prepare yourself!
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by expanding my intrusion detection but
it still stops every few days or so. I have a cron job that tests for
it and restarts it when necessary.
Anyone else have experience on non-Linux systems?
Cheers.
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Ron Wheeler
President
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email: rwhee...@artifact-software.com
skype
part...
I don´t know...
one is
siptrunk.ovh.net
and the other one is
sip.ovh.fr
how can i determine and how could that affect... I mean... why do they
interfere at all?
thanks,
yves
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Davis Drive NW · Huntsville, AL 35806 · US
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Check us out at: http://digium.com · http://asterisk.org
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/
On the trampoline of life's experiences, Striving towards a saintly
life in the midst of these materialistic turbulences.
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for the following:
- good quality
- fast turnaround - can read and understand a script and get it right
the first time
- ability to find the talent again if you need re-recording.
- neutral accent
Ron
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go at the top group and
would vote for a change in the rules.
And do not even start on the Chevy vs. Ford debate respecting the technical
superiority of Pine over Outlook. GAWD... Life its too short as it is.
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email: rwhee...@artifact
as a whole, this is where your reply goes)
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posting.
In the long run.
--Don
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I've got several stable systems in production, I don't mess
with them much.
Doug
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active without any
problem before messing with the update)..
What should the clever sysadmin do?
Thanks in advance,
Aldo
Try:
# standard asterisk command-line. No verbosity
strace -eopen asterisk -U asterisk -c
See which module was the one last loaded.
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Ron Wheeler
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see call drops, delay in responding, time out,
re-transmission of SIP messages. If load is reduced, it settles again
to normal.
*Thanks Regards,*
Amit Patkar
On 1/28/2014 12:32 AM, Ron Wheeler wrote:
Can you get a reading of the total number of I/Os during your test?
Peak IOPS?
That might
be better able to verify that.
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be
so low,
that it is not worth using something like fail2ban to override them? Food
for thought!
murf
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phone: 866-970-2435
:04:51 sip32 kernel: asterisk[2066] trap invalid opcode
ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000]
Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode
ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000]
Kind regards,
Jonas.
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.x86_64 #1 SMP Wed Oct 16
18:37:12 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux
Jonas.
On 20-11-13 14:11, Ron Wheeler wrote:
Is it possible that in your build you mixed 32 bit and 64 bit libraries?
Ron
On 20/11/2013 8:06 AM, Jonas Kellens wrote:
Hello,
I have installed asterisk 1.8.24 (from source
On 28/10/2013 4:12 PM, Mark Wiater wrote:
On 10/28/2013 3:59 PM, Ron Wheeler said:
I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the
Asterisk - No analogue.
I don't have any problems
mailto:emik...@rimmkaufman.com
http://www.rimmkaufman.com
http://twitter.com/rimmkaufman
http://www.linkedin.com/company/85385
http://plus.google.com/104980442218952272663/posts
http://www.facebook.com/rimmkaufman http://www.RKGblog.com
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email
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] What linux distro most popular for Asterisk
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or update options visit:
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webinar every Thurs:
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dbpass = MyPassword
dbport = 3306
dbsock = /tmp/mysql.sock
dbcharset = latin1
requirements = warn
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in asterisk for i see the SQL query ?
2013/6/4 Olivier CALVANO o.calv...@gmail.com
mailto:o.calv...@gmail.com
oh ron thanks for your help :
We have deleted all commented line, only put the configuration and
now that's work !
2013/6/3 Ron Wheeler rwhee...@artifact
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/listinfo/asterisk-users
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is to be
automated on a small system with no GUI, only console/SSH access. As such,
running a full featured audio recording/mixing application in realtime (with
user control) is not an option. :/
--Tim
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be best advised to request (paid)
assistance with the project on the [asterisk-biz] list.
Kind regards,
Chris
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code ends up unredistributable. (But it works as well as anything). Then
write a Web app on the database server to display wanted CDR entries.
What about a script to convert the CSV to HTML and ftp the html file to
a web server where it can be accessed as a browser page?
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On 23/04/2013 11:42 AM, aristidis tsitras wrote:
On 04/23/2013 06:23 PM, Ron Wheeler wrote:
On 23/04/2013 11:09 AM, A J Stiles wrote:
On Tuesday 23 April 2013, aristidis tsitras wrote:
Hi. i am running asterisk in a low powered machine (alix2d13 from
pcengines) without any gui. the machine
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New to Asterisk
as an artist.:-)
Ron
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New
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list
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Thurs:
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+1
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said, given the number of Asterisk installations being
installed each day by first-time administrators, the traffic here seems
pretty reasonable both in volume and in level of difficulty.
Ron
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Ron Wheeler
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email: rwhee...@artifact-software.com
skype
and tries to dispatch the call.
This makes it hard to carry on a conversation.
Ron
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On 02/01/2013 3:33 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Wednesday, January 02, 2013 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
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On 30/12/2012 11:13 AM, Patrick Lists wrote:
On 12/30/2012 04:26 PM, Ron Wheeler wrote:
I participate in a lot of lists and top posting is now the norm since
people want to see quickly if the message is worth reading.
Isn't it a bit of a stretch to extrapolate your experience with your
introductory webinar every Thurs:
http://www.asterisk.org/hello
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list
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Excellent.
It appears that Getting Started has a lot more stuff in it than the
documentation for 1.8.
Very helpful.
Ron
On 29/11/2012 12:31 PM, David M. Lee wrote:
On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:
That is a good answer.
Thanks.
Any reason why it is not documented?
It's
picks up but the incoming caller is not
connected and still hears the pbx ringing the SIP phone even though the
SIP phone is no longer actually ringing.
Ron
On 28/11/2012 1:15 PM, Joshua Colp wrote:
Ron Wheeler wrote:
I have 2 analog trunks.
They answer the incoming call, do the welcome
is
working on SIP.
Ron
On Sat, Nov 24, 2012 at 4:51 PM, Ron Wheeler
rwhee...@artifact-software.com
mailto:rwhee...@artifact-software.com wrote:
I looking through my logs, I found that people where probing my
SIP accounts looking for passwords.
Asterisk was helping them out
I had to install fail2ban and configure it to watch Asterisk.
Ron
On 27/11/2012 2:11 PM, Mitul Limbani wrote:
You might want to share the know how over here if its not a chan_sip
patch.
Mitul
On Nov 28, 2012 12:28 AM, Ron Wheeler
rwhee...@artifact-software.com
mailto:rwhee...@artifact
: == Spawn extension
(voicemenu-home, 102, 1) exited non-zero on 'DAHDI/2-1'
[2012-11-27 14:45:42] VERBOSE[3589] sig_analog.c: -- Hanging up on
'DAHDI/2-1'
[2012-11-27 14:45:42] VERBOSE[3589] chan_dahdi.c: -- Hungup 'DAHDI/2-1'
(END)
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email
to someone's clear
recipe, I was able to get it working.
I hope that this can be worked into a release soon.
Ron
--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
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