Have you tried using the EVAL function?
On Tue, Jan 24, 2023, 7:38 PM
wrote:
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> asterisk-users@lists.digium.com
>
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Linux x.y.com 3.10.0-693.5.2.el7.x86_64 #1 SMP Fri Oct 20 20:32:50 UTC
2017 x86_64 x86_64 x86_64 GNU/Linux
I try to keep up with the latest versions of everything.
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
alls could be impacted by
this. It seems clear that the ability to make calls to customers and
suppliers will become uncertain and potentially vary from cases to case.
Ron
On 16/12/2017 1:05 PM, Eric Klein wrote:
Hi Ron
There was an article back in July looking at what might happen
How does th
traffic crosses many networks.
Am I way off track?
Ron
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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
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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.
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:
Hello,
I'm used to install Asterisk on Debian stable platforms.
A customer is asking how I would proceed on a CentOS platform
. You can likely specify
what you want but you need to look at the caller Id setting on the
Extension setting form.
It looks like you are trying to tie extensions to trunks directly - No
press 1 for Ron, 2 for Paul, 3 for tech support, etc.
You want incoming calls to number xxx- to go
be able to help you.
Surely, you mean the Biz List
Doug
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version of Zoiper on my Android cell as a production
phone but you can use that to test each extension as you set it up.
The IVR setup is pretty straightforward.
Are there any potential issues that are of particular concern. Ring
groups, IVR menu design?
Ron
On 12/12/2017 10:30 AM, basti
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.
I hope that this extra info helps find a solution.
Ron
On 12/12/2017 3:41 AM, basti wrote:
Size:
- one location
- 15 IP
do you want to deal with - one man shop with a
genius in charge that you may only be able to reach after hours or a
shop with techs of various skill levels that can give you a believable SLA.
Ron
On 11/12/2017 3:53 PM, basti wrote:
Hello,
we plan to move a PBX to asterisk and searching
? I have never tried
anything like this.
Perhaps if you clarify this, someone might have a suggestion.
Ron
On 11/12/2017 9:16 AM, Tech Support wrote:
Hello;
I certainly appreciate your response. In fact, I used that exact
solution for three of the incoming lines. I setup ring groups an
.
https://www.freepbx.org/ring-group-and-follow-me-ring-strategies-1-of-2/
might be useful.
Ron
On 08/12/2017 2:17 PM, Tech Support wrote:
All;
I have an interesting scenario where I have a small office with
maybe half a dozen phones and several incoming lines. The calls are
routed
Great.
Let me know how your policy works out.
I would not mind trying it myself.
I have no intrinsic objection to doing things the right way but
sometimes one just needs to get the phones working!
Ron
On 15/03/2017 4:06 PM, Dan Cropp wrote:
Thank you Jason
After following your steps
version of CentOS 7
Ron
On 15/03/2017 12:40 PM, Dan Cropp wrote:
Thanks Jason.
I will try to explain what I’m seeing for this issue.
I did a fresh install of CentOS 7 Minimal into a VM with VMWare
Workstation. Followed the Asterisk from Source instructions using
pjproject 2.6 and asterisk
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
want to look elsewhere.
It seems that a lot of things do not work with Selinux or
have no instructions about how to make them work with Selinux that it
almost seems like a useless feature.
Ron
On 14/03/2017 2:21 PM, Tzafrir Cohen wrote:
On Tue, Mar 14, 2017 at 06:03:33PM +0100, Jean Aunis
I have FreePBX 14.0.1beta20 running on Centos 7.3.
What problems are you having?
The latest emails don't have any details about the problem or what you
have tried.
Ron
On 14/03/2017 2:21 PM, Tzafrir Cohen wrote:
On Tue, Mar 14, 2017 at 06:03:33PM +0100, Jean Aunis wrote:
Hello,
Did you
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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email: rwhee...@artifact-software.com
(using a second NIC) that should help performance.
Is the SIP network on the same network as your internet/data LAN?
Ron
On 04/01/2016 1:15 PM, IPN Comm wrote:
I was wondering if anyone can give me any pointers or insights of
whether or not to have an asterisk server behind a firewall.
I have
have Centos 6 with MySQL 5.1.73 and PHP 5.3.3 with FreePBX 2.11.
Ron
--
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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
the same network even
with two way video active or during multi-party conferences (mix of
Skype and telephones in the group).
I would like to have a reliable 2 way conversation using Asterisk but
have not found any suggestions about the source of the problem or how to
fix it.
Ron
On 12/03/2015
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
Your characterization may be true but Skype works much better than SIP
when it comes to sound quality.
I have SIP softphone with Asterisk server and Skype on the same
workstation.
Skype just works better over the same network.
Ron
On 12/03/2015 9:26 AM, A J Stiles wrote:
On Thursday 12 Mar
Have you done the math for the network connections? BTF and external
What bit rates for the sound?
What codecs?
How are calls coming in - SIP - analogue
Disks OK(low IO per second)? Caching working OK?
CPU may not be the problem if your CPU utilization is really that low.
Ron
On 02/03/2015 10
with the main
firewall/router and a virtual host.
Firewall is now in production but it was a bit of a learning curve for me.
There are big differences between 6 and 7 and I would let some other
Asterisk users switch before going to 7.
Free advice and worth every cent!
Ron
On 12/02/2015 9:25
proceeding normally.
Unchecking this box solved the problem.
It may not be related to your problem but if it is the cause, you will
spend a lot of time trying to fix this in Asterisk. :-D At least I did!
On the bright side, it does force people to get point in a hurry!
Ron
On 22/11/2014 12:50
will get different people helping you
depending on the question that you ask. We are not all experts at
everything. Most of us are people like you or end-users who are
supporting their own company's phone system.
Ron
On 08/10/2014 12:34 AM, Dania Asi wrote:
Dear Mr. Adam,
Thank you for you
, how many simultaneous calls can you support?
Just to be sure that recording is the issue.
Ron
On 23/07/2014 4:29 PM, Scott Griepentrog wrote:
Your bottleneck is most likely your drive bandwidth. Even with SAS
drives, you'll need to move to a raid 5+ solution with 6+ drives to
continue
Do the calculations for both and see what the answer is.
The nice thing about having a model is that you can test configurations
without actually having to build one until you are confident that it
should work.
Ron
On 23/07/2014 5:04 PM, Eduardo Leones wrote:
Thanks for the feedback
/
On the trampoline of life's experiences, Striving towards a saintly
life in the midst of these materialistic turbulences.
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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
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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President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
systems replaced.
Ron
On 14/03/2014 9:52 AM, James B. Byrne wrote:
On Thu, March 13, 2014 15:32, Kevin Larsen wrote:
On 13/3/14 6:27 pm, Eric Wieling wrote:
This is an example of why I top post. Who wrote what?
+1-1 = 0
I do not care about where people put their replies so long as I can
as a whole, this is where your reply goes)
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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
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posting.
In the long run.
--Don
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email: rwhee...@artifact-software.com
skype: ronaldmwheeler
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result in forum discussions.
Ron
On 02/03/2014 7:44 PM, Doug Lytle wrote:
Stefan Gofferje wrote:
Now it's
hardly 50 new mails per week.
Is the list dead? Or is the project dead?
It's called being a mature project. And, I don't call averaging 400
messages a month as being a dead list.
And, once
DAHDI might be the culprit.
You may have had a better version from Asterisk than the new one that
YUM got you.
Check to see if YUM gave you a new DAHDI. Who's your daddy now?
You may want to rebuild the Asterisk DAHDI and install it over the DAHDI
from your Linux distro.
Ron
On 12/02
the bits stored
for each second of audio?
What happens when you do this?
Ron
On 29/01/2014 7:34 AM, Amit wrote:
Thanks Ron.
I will try to get these readings. About RAM disk, I will study on how
to create RAM disk and conduct this test again.
There is no bottleneck on network.
After 80 calls, I
.
What is your network capacity? Usually one can write faster than the
network can deliver - just to make sure that you are chasing the right
bottleneck.
What happens at 80 calls to tell you that you have run out of IOPS?
Sorry for more questions than answers.
Ron
On 25/01/2014 12:26 AM, Amit
own
legitimate phones.
Ron
On 19/01/2014 9:40 AM, Steve Murphy wrote:
On Sat, Jan 18, 2014 at 3:59 PM, Steve Edwards
asterisk@sedwards.com mailto:asterisk@sedwards.com wrote:
On Sat, 18 Jan 2014, Jerry Geis wrote:
I see MANY of these in my log files:
[Jan 15
flexibility in adding power to your setup
at an Amazon.
I guess that one can decide what are the critical points that need to be
tested (call volume, call quality, user connectivity) and devise a test
setup.
Ron
On 22/11/2013 1:18 PM, Todd R. wrote:
I would have said the same thing a while
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) but I can not start up
Asterisk :
[root@sip32 admin]# /usr/sbin/asterisk -r
Illegal instruction
[root@sip32 admin
with 64
bit applications.
Ron
On 20/11/2013 8:15 AM, Jonas Kellens wrote:
Hello,
how can I mix libraries ?
I have installed prerequisites from yum and asterisk from source (make
make install).
My kernel :
[root@sip32 asterisk-1.8.24.0]# uname -a
Linux sip32.domain.tld 2.6.32-358.23.2.el6
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
Ron
On 28/10/2013 2:29 PM, Eddie Mikell wrote:
All,
The users in our organization are well, quite frankly, sick of phone
service that is being provided. The choppy phone calls, and drop outs
are detrimental to our sales force.
I've tried about everything I can think of.
Moved
Hard to give an answer but for us it is
Centos - 1, others - 0
Ron
On 17/10/2013 6:16 AM, emilianovazq...@gmail.com wrote:
Most tutorials over internet are based on Centos and Ubuntu.
Centos is the base distro of FreePBX, Elastix and Trixbox and always have a lot
of users.
I use ubuntu
do if they attacked these ports.
Ron
On 18/09/2013 2:29 PM, Ira wrote:
Re: [asterisk-users] RTP port ranges Hello Thorsten,
Tuesday, September 17, 2013, 1:05:15 AM, you wrote:
Where is it stated that you MUST use 1-2 ???
Someone else please ?
Well, I don't use that range
will be happy to use it.
yum is set up to have a number of repos configured and a local one
is just fine.
Ron
OS CentOS 6.4
Asterisk version 1.8.13.0 11.4
$ find / -name *lua
I suspected that the restriction might
be policy rather than technical.
Is there anything that guides the loading of software via USB or
DVDs on isolated machines or is my suggestion about a local yum
repo, a workable solution?
Ron
On 18
It looks like your database configuration is missing in Asterisk.
It is making up information about the connection using defaault values
as if it did not find any database configuration.
Ron
On 03/06/2013 10:49 AM, Olivier CALVANO wrote:
Hi
i have installed a new Asterisk server on Fedora
that they know about.
Ron
On 03/06/2013 12:19 PM, Olivier CALVANO wrote:
No other idea ?
2013/6/3 Olivier CALVANO o.calv...@gmail.com
mailto:o.calv...@gmail.com
Hi
i have installed a new Asterisk server on Fedora. My first server
use Asterisk 1.6.x with a MySQL CDR
Do you have this problem in your conf file?
http://forums.digium.com/viewtopic.php?p=63736
The parser won't accept an ; (semicolon) for remarks! So he found at
the first the old remarks and tried to access my database with the false
data.
Ron
On 03/06/2013 3:18 PM, Olivier CALVANO wrote
Well, at least you are making progress.
What is the error in the debug log?
Ron
On 03/06/2013 8:03 PM, Olivier CALVANO wrote:
grrr no in asterisk -d i have no error, but when i start normaly
asterisk i have :
[Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime:
Failed
What are you trying to accomplish?
What is the USB 'sound card' attached to?
Your description is too cryptic for someone to propose a solution.
Ron
On 28/05/2013 12:45 PM, Tim Nelson wrote:
Greetings-
I've got a curious project that I could use some input on. I'd like to use
Asterisk
://www.asterisk.org/hello
asterisk-users mailing list
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phone: 866-970-2435, ext 102
Sorry for the blank message. Fingers pressed send while brain was disenaged.
Would Audacity be a better choice?
http://wiki.audacityteam.org/wiki/Multichannel_Recording
Ron
On 28/05/2013 1:23 PM, Tim Nelson wrote:
- Original Message -
What are you trying to accomplish?
What
-mail, upload, whatever).
Sounds like a security monitoring package (minus the video) should do
the job?
A little Googleing shows up these.
http://oreka.sourceforge.net/about/
http://www.ubuntugeek.com/how-to-recording-internal-audio-in-ubuntu.html
What else do you want it to do?
Ron
On 28/05
the election.
If you hire a company outside your country to do this, you can make it
hard to detect and impossible to prosecute.
Ron
On 23/05/2013 3:40 PM, cjwstudios wrote:
As long as you're dialing a screened registered voter list and don't
call .gov or .edu, you're fine.
On Wed, May 22, 2013 at 5
Good comment.
Another feature suggestion
You might to ask the person to press 1 to confirm or 2 to leave a
message if the appointment is not going to be kept or 0 to reach the
receptionist to reschedule the appointment.
Ron
On 26/04/2013 7:06 AM, Chris Bagnall wrote:
On 26/4/13 10:38 am, jg
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?
--
Ron Wheeler
President
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
http://www.artifact-software.com/?page_id=1666
Would this help?
Put a JasperReport graph or two in a report step.
Ron
On 10/04/2013 2:02 PM, Steve Edwards wrote:
On Wed, 10 Apr 2013, Carlos Alvarez wrote:
Is anyone using something to log SIP results (connected/not, latency)
that they really
as an artist.:-)
Ron
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email: rwhee...@artifact-software.com
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New
requirement is coming from.
http://www.artifact-software.com/?page_id=929 is the website link if you
want more info.
A short brochure is available. If anyone wants one, please contact me
off-list.
Ron
On 11/01/2013 5:22 AM, Olivier wrote:
Hi,
I would like to edit reports showing how fast operator
asterisk-users mailing list
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list
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solutions here, deserve
whatever gets thrown at them.
Ron
On 10/01/2013 5:32 PM, C. Savinovich wrote:
Isn't this precisely the raison d'être for [asterisk-biz]?
Oh my goodness!, the asteriz-biz? nooo, they will kill you if you try
to post anything offering your services!... that list ceased
their services.
Whether they get our e-mail here or from our web-sites or from our
business cards, we will get approached.
We are free to buy or not to buy. Read or ignore.
In a forum like this, that is almost the only price of free advice.
Ron
On Thu, Jan 10, 2013 at 5:32 PM, C. Savinovich
for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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+1
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email: rwhee...@artifact
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
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype
and tries to dispatch the call.
This makes it hard to carry on a conversation.
Ron
--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
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
it easy to figure out the history if you
were not following the discussion closely.
Ron
On 29/12/2012 10:02 PM, Logan Bibby wrote:
I suppose I'm one of the few people that remember the content of
threads by subject and easily catch up...
I'm also on my phone 99% of the time time and the way
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
answers, your participation will be encouraged.
It is only long discussions that will miss your input.
Ron
/Benny
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I just wish that my biggest problem with Asterisk was top or bottom posting!
Ron
On 30/12/2012 9:45 PM, James Mortensen wrote:
Sorry for double posting, but I realized it was JIRA I spoke with
Digium about, not Google Groups and the mailing list... However, I do
think it's worth investigating
.
That is the day you box up all the crap you got and exchange it for what
you really wanted.
It is a religious holiday in the old British Commonwealth (probably
Scottish in origin).
Ron
But anyway the best way to test time-based rules is on a VM that has a
copy of your configs, and just change the time
It seems like a safe thing to do.
You could also ask about the impact of making an existing column a
primary key, in a MySQL forum.
Leandro's solution seems to be a good one as well and does guarantee
uniqueness.
Ron
On 06/12/2012 12:25 PM, Leandro Dardini wrote:
Yes, go for it. However
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asterisk-users mailing list
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That is a good answer.
Thanks.
Any reason why it is not documented?
Ron
On 29/11/2012 11:52 AM, Mikhail Lischuk wrote:
Shitian Long wrote 29.11.2012 18:40:
There is a part of dial plan from sample extension.conf above. My
Question is how same = key word works .
Thanks
same is used
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
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email: rwhee...@artifact-software.com
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phone: 866-970-2435, ext 102
on hold
; instead of ringtone
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as to what could be causing this?
Bill Dunn
Try setting:
relaxdtmf=yes
We used to have that same problem on most of our servers. Setting
relaxdtmf to yes solved the problem for us.
--
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc
Bill Dunn - VCI Internet Services wrote:
Thanks Ron. I have had my chan_dahdi.conf file set as follows with the
same
result.
[trunkgroups]
[channels]
switchtype=national
usecallerid=yes
callerid=asreceived
cidsignalling=smdi
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
:
Greetings Ron-
Just wanted to give you a heads up about an alternative SCCP channel
driver available for Asterisk. Please see here:
http://freecode.com/projects/chan-sccp-b
I have no experience with it (nor SCCP in general) but just wanted to give
you an option in the event the included SCCP
Is the chan-sccp-b project the same one that got put in SVN of 1.8 branch?
I have not been able to find anything definitive that says so, I really
need 1.8 branch so trying to see which is the best way to go.
Thanks
On Thu, Jun 14, 2012 at 9:34 AM, Ron McCarthy ronmc...@gmail.com wrote:
Hello
Hi List,
Has anyone been running SCCP with a larger number of phones? Im looking to
deploy like 75+ phones and I want to keep SCCP so I don't have to upgrade
them and for the SLA, some phones also have no SIP software for them so im
forced to keep SCCP. Does anyone have any experience with this?
($agi, 'new_years_day', '10to7');
$agi-hangup();
}
else {
$agi-exec('Goto', 'welcome');
}
}
--
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc.
--
_
-- Bandwidth and Colocation
themselves.
Ron Bergin
Network Operations Administrator
Fry's Electronics Inc.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
Jason Parker wrote:
On 03/06/2012 12:31 PM, Ron Bergin wrote:
Mathew,
Each of those odbc modules are unavailable i.e., marked with XXX
I even deleted the asterisk build directory and started over, but had
the
same results.
What prereqs do I need besides these:
mysql.i386
-2.6.0
libpri-1.4.12
What am I missing?
--
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Paul Belanger wrote:
On 12-03-06 12:05 PM, Ron Bergin wrote:
However, I'm getting a seg fault error when
starting asterisk.
# /usr/sbin/safe_asterisk: line 145: 27014 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} 21
Matthew Jordan wrote:
- Original Message -
From: Ron Bergin r...@i.frys.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, March 6, 2012 11:05:35 AM
Subject: [asterisk-users] Compiling asterisk with mysql support
When I run 'make memuselect', I select res_config_mysql
Matthew Jordan wrote:
- Original Message -
From: Ron Bergin r...@i.frys.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, March 6, 2012 11:05:35 AM
Subject: [asterisk-users] Compiling asterisk with mysql support
I have a working asterisk test box that I'm rebuilding
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