There are much better solutions than doing a RAM drive. While it may
be stable (not in my experience, I advise using different servers for
different tasks (with redundancy obviously). A phone switch should be
just that, a recording server should also be just that (in demanding
I still hope someone would enlighten us by his experience in doing call
recordings without recording to RAM Drive.
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i can only think of an asterisk box the right dialplan.
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he's capturing the audio at the network layer
i'd better stay with my 3Gigs RAM drive
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Vicidial has call center / CRM integration with asterisk ... with many years
of bug reporting ... it open source.
On Sat, Apr 26, 2008 at 12:11 PM, Kashif Naeem [EMAIL PROTECTED]
wrote:
Hello All,
A company has two requirements:
1) They are looking to develop its own CRM
2) Second thing is
I am facing the problem that
zaptel is not going online when booting,
most people run it from /etc/rc.local ...
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Install kernel-version.src.rpm with the following command:
rpm -Uvh kernel-version.src.rpm
this is from http://fedoraproject.org/wiki/Docs/CustomKernel
ubuntu is better/safer/faster/has 5 year of updates ... you name it.
On Sat, Apr 26, 2008 at 4:31 PM, bilal ghayyad [EMAIL PROTECTED]
I thought most people ran it from /etc/rc.d/init.d/zaptel
here is what README file says :
Installation
Note: If using `sudo` to build/install, you may need to add /sbin to
your PATH.
--
make
make install
# To install init scripts and
By what criteria do you form the opinion that Ubuntu is better, safer, and
faster?
well I don't know what's your favourite but compared to fedora ... i guess
it is.
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By what criteria
its a like a car you've got to drive it to feel it.. so give it a shot
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if wardialer is the correct term
this must be a predictive dialer ... which is simply a dialer that dials a
list of numbers you supply to him ... then you need to configure if for when
to pass the call to you when to hung up ... vicidial is a good project to
start with
D is for Disturbing other poeple.
On Sat, Apr 26, 2008 at 10:39 PM, Benny Amorsen
[EMAIL PROTECTED][EMAIL PROTECTED]
wrote:
Steve Totaro [EMAIL PROTECTED] writes:
But then that gets back to my Intel C2D show as two procs. 2 x 2 = 2.
Or is C2D not four cores?
D is for duo.
/Benny
you saying :
You make it sound as if only one person would be dialing one number at
a time
he's saying
but I want to be
on the line and if possible complete / talk on certain calls.
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some people use a war dialer to provide call centers with numbers for
their campaigns ... if number called rings the number is valid if it doesn't
its invalid discarded. i wonder if that is legal .. its basically a scan
of the network for valid numbers (that is potential buyers).
i once was
I will be releasing a dialer under the GPL shortly that has a
very low profile, AJAX web interface, and will be able to do just what
you want.
what programming language will you use under the hood ?
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can tell you that it is an actual asterisk application (with a conf
file) and html for the web interface (and obviously a database).
I'd like to start something similar with python/Mysql under the hood a web
page at the front end (ajax) ... but so far my time does not help. I wonder
if there
exten = wardial,1,NoOp(wardialing:
i didn't mean i wanted a wardialer ! i meant a simple predictive dialer
(that is no rich features, only to be able to make transfers internaly
externaly to show data of called lead to agent)
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Please contact me off-list if you would like to test out my beta
i'd be glad to do that.
It is connected to a TDM PRI. Where are you calling? I pay a penny a
minute but would be glad to eat that cost for real world testing.
I also use PRI links only ... the thought of testing a
http://www.soft-switch.org/unicall/mfcr2/ch02.html
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Make sure you get a helpful tech on the phone. Many times they will
just dismiss you with we cannot do that even though they may be able
to.
i always say if you pay your bills you should get the support you diserve.
every provider is almost always willing to help out his clients if they
i suggest a look at digium's hardware compatibility list on thier website.
i would also worry about concurrent calls thus concurrent recordings, 48
with your actual card which i guess is acceptable load to your hardware but
i am not an expert in this.
recording to disk (even scsi ones) will make
I can dial out to other numbers without issue.
Calling the number from a separate PSTN phone works fine.
I have had such problem last week. from PRI interface (i get busy tone as
fast as i finish typing last digit hit dial) from analog interface I can
make the call without problem.
my
vicidial ... vicidialNOW
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is registerattempts=0
in your sip.conf ?
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Do you plan to use it in a call center or casual business office ?
Call Center
I still think your best bet is vicidial.
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Wow, what a disaster of an open source project. Install docs
are impossible to use. Many, many inaccuracies.
i think you just need someone set it up for you ... think of it as an air
conditionning system, you can use it but can never install it on your own
unless you're from the field.
No. The problem is the docs are all wrong on Covide's project.
The web site says one thing, the readme another. Neither are correct.
well you may be correct but we must admit one thing, it takes a lot of
dedication to start continue a real project ... and only for that every
developper
help would be greatly appreciated. Is there anyone with a
similar setup to this that has any suggestions/tips?
Thanks,
Jacob
Jacob Arthur, MCP
ATS
[EMAIL PROTECTED]
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Hello list,
I have a customer who is interested in standardizing on dell servers for
asterisk deployments.
Has anyone had success with a particular configuration?
Anything specifically to watch out for?
Thank you for your time,
Art
Arthur Miller
Sr. Sales Associate
VoIP
The Digium cards are known to steal IRQ's.
The Sangoma cards do not.
Arthur Miller
Sr. Sales Associate
VoIP Supply, LLC.
454 Sonwil Drive
Buffalo, NY 14225
716-250-3871 OFFICE
716-630-1548 FAX
[EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED]
NOTICE: The information
Hi John,
We've had similiar issues with customers behind the 2920 connecting to a hosted
asterisk system. If you rebooted a phone it often didn't re-register, Checking
the NAT sessions table on the router revealed stale nat sessions open for the
phone.
On the advice of Dreytek we found a fix
Hi Arlen,
I'm interested in seeing what setup you settled on to get decent voice quality
over the Sat link? Which codec are you using, and what is the bandwidth usage?.
Are you doing just one concurrent call, Or multiple?.
-
Regards,
AJ Stanfield
- Original Message -
From: Arlen
Hi Gilles,
You can't tunnel UDP through SSH.
Some of the newer Grandstream handsets support OpenVPN and are a bit cheaper
than the Snom alternatives.
-
Regards,
AJ Stanfield
t: 0161-850-4001
e: a...@dmcip.com
w: http://www.dmcip.com
- Original Message -
From: Gilles
- Original Message -
From: A J Stiles asterisk_l...@earthshod.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, 4 April, 2012 10:48:02 AM
Subject: Re: [asterisk-users] cross ivr is comming in my ivr system
On Wednesday 04
Hi Anam,
Hope this helps explain Asterisk version numbering:
http://leifmadsen.wordpress.com/2011/08/29/asterisk-10-asterisk-1-hh10/
Easy to get confused!.
Cheers,
AJ.
- Original Message -
From: Satria Anamarta anam.satri...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial
Hi Binni,
We run a number of Asterisk servers on virtual machines. I'm not heavily
involved in the virtualisation side of the business so i'm afraid i can't give
you much advice on it, Past saying it is possible to have an Asterisk System up
and running reliably on virtual machines.
Our
Boo, and i felt so special for a few minutes this morning! :(
--
AJ
[YOUR AD HERE]
- Original Message -
From: Steven Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 27 April, 2012 9:28:18 AM
We've just had one of each delivered for us to play with in our lab (Literally
an hour ago!). Not had chance to play with them yet, But initial thoughts are
they look good. Build quality seems fine for the price. I'll form more of an
opinion when i get chance to play with them properly
Log into the Linksys GUI, Look at what SIP account it is registering to
asterisk with then run sip show users ?
- Original Message -
From: Hassan Abdalla hagga...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Saturday, 26 May, 2012 6:08:59 PM
Subject: [asterisk-users] Linksys PAPT2
Hi. I can't figure this one out. Hope someone can help me.
[EMAIL PROTECTED]:# cat /etc/odbc.ini
[Asterisk]
Description=PostgreSQL asterisk
Driver=PostgreSQL
Trace=No
TraceFile=/tmp/odbc.log
Database=asterisk
ServerName=localhost
UserName=
Password=
Port=5432
Protocol=7.4
Sorry about this. Just figured it out.
In res_odbc.conf its supposed to be pre-connect and not preconnect.
On Saturday 01 January 2005 02:30, Arthur B Olsen wrote:
Hi. I can't figure this one out. Hope someone can help me.
[EMAIL PROTECTED]:# cat /etc/odbc.ini
[Asterisk]
Description
/etc/group
/etc/passwd
/etc/shadow
The line looks like a yp line. It tells the pam module to search the NIS
server for users, groups and password
On Saturday 08 January 2005 05:28, Michael Levenson wrote:
Can someone help me answer this question?
Where would you most likely find a file with
I think you need that provisioning tool from digium. And you need a unix
system to compile and run it. I dont think theres any port to your OS.
Sorry...
On Saturday 08 January 2005 08:08, Daiku wrote:
Hi,
hoping that experienced hands will quickly show me the right way: after a
fruitless
I though that the externip was used within the sip communications, so it is
sent as is, and resolved on the other side.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dennis Cartier
Sent: Wednesday, July 14, 2004 9:51 AM
To: [EMAIL PROTECTED]
Subject:
I have also been having problems today registering... I contacted them, but
they have no known issues. It finally did register on it's own.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre Normandin
Sent: Wednesday, July 21, 2004 8:44 PM
To:
Hope this is the right maillinglist.
I would like to know how i can secure the iaxy. Or is the the sad truth that
anyone with an iaxyprov program can change any box not behind a firewall?
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So i guess were screwed. These unusable thingies are quite expensive.
Good thing i only bought two.
Now im really nervous about the isdn pri card i bought. Gonna try it out
tonight. Hope its different.
Is the software for iaxy open source. Then maby it can be fixed.
On Wednesday 22 December
But whats the future of iaxy. Are these problems being fixed. Or is the whole
project dropped?
On Thursday 23 December 2004 01:38, Kristian Kielhofner wrote:
Arthur B Olsen wrote:
So i guess were screwed. These unusable thingies are quite expensive.
Good thing i only bought two.
Now im
Got a really wierd problem her. Maby it's a bug.
But before i report it, i'll try my luck here.
I have one asterisk server on public ip.
I have two identical hardphones on two different LAN's. The firewall are
different.
Both are configured in asterisk with nat=yes and qualify=yes.
For one
Hi Ish,
I assume you are using Fail2Ban to monitor the logs for dictionary attacks - If
so, the following regex should work for 1.8:
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No matching peer
found
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