Have you looked at the privacymanager function in Asterisk?
PaulH
On Wed, 2008-03-19 at 10:31 +0530, Janu Mukherjee wrote:
Hi,
I have our software with SIP running on it.I configured asterisk
server as proxy. How do I implement the call screening
features(incoming and outgoing) using
I was trying to find out how I could put in a greeting when a caller
***first*** joins the queue.
I searched high and low but could only find (in queues.conf):
. announce, which is announcement to the agent
. announce-frequency which is announcement of queue position
.
James Lamanna wrote:
Hi,
Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone?
I've tried http://[ip address]/admin/spacfg.xml however that file
doesn't appear to exist.
Yes it is. It requires a 2 step process.
1. Configure the Provisioning Parameter Report Rule to
I would think you'll need to do a Playback() of this message before the
caller enters the queue, as I'm not aware of such an option provided by
app_queue.
Exten=100,1,Answer()
Exten=100,n,Playback(greetings-earthling)
Exten=100,n,Queue(xyzqueue)
Exten=100,n,Hangup
-Original Message-
I would think you'll need to do a Playback() of this message before
the
caller enters the queue, as I'm not aware of such an option provided
by
app_queue.
Exten=100,1,Answer()
Exten=100,n,Playback(greetings-earthling)
Exten=100,n,Queue(xyzqueue)
Exten=100,n,Hangup
Thanks Mark for your
What kind of information are you looking for? configuration or? If you
look in our manuals our cards and the Digium cards configure the same
in zaptel and zapata.
Hi James, I have purchased a CB24-FXS-UNIV and am using a TE412P card
with a Asterisk 1.4 box.
One port of the card is connected to
I've been using Druid since their V since their V.3.2.5, and I have to say I
love it. I haven't done a lot of advanced things with it though, but I know
that their new version has almost all of it. Upgrade is not an option, a new
license is, which is why I'm still with V325.
Either way, Druid is
Lee,
I'm pretty sure you can using macros and what not. Unfortunately, I'm not
that experienced to comment on it, but I'm positive that can be done with
one of those if, else things.
Mark.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee, John
Check the number of calls waiting in the queue, then play the message if
more than 0
example code (written in the TBird IDE)
Exten = 100,1,Answer()
Exten = 100,n,Set(NumWaiting=${QUEUE_WAITING_COUNT(${QUEUENAME})})
Exten = 100,n,GotoIf($[${NumWaiting} = 0]?JoinQueue)
Exten =
On Tue, 18 Mar 2008 14:50:52 + (GMT), Gordon Henderson
[EMAIL PROTECTED] wrote:
BT nearly always try to sell featureline on business lines these days.
Would sir like a 3 of 5 year feature line contract?
Fair point, Gordon.
But in their defence, I got free installation for Featureline
On Sun, 25 Mar 2007, Gordon Henderson wrote:
and change the Optional Rule to:
3,3,7,1,0;10,4,7,2,0;60
someone correct me if I've goofed!
Well, it's been a year since I wrote this, and of-course I goofed and
no-one corrected me ;-)
Bother.
So I've looked again.
In the UK, We Spring
On Tue, 18 Mar 2008 14:06:44 +, Paul Goodyear [EMAIL PROTECTED]
wrote:
Hi,
I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there
are 3 BT lines connected directly to these ports.
I've just had a quick Google for you and at least one person solved
this by replacing the cable
Alex Balashov wrote:
Hello List,
I'm using a dialstring like the one below. I want to have three
different things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(2[:5000][:5000]))
These 3 things could happen:
1, Caller hangs up
2, Callee hangs up
3, The 20 seconds
Isasterisk-1.4-current.tar.gz(13-Mar-2008 15:06 11M) not the same
as asterisk-1.4.18.1.tar.gz (18-Mar-2008 12:24 11M ) ?
Should be?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
On Wed, Mar 19, 2008 at 09:35:45AM +0100, Adrià Vidal wrote:
Isasterisk-1.4-current.tar.gz(13-Mar-2008 15:06 11M) not the same
as asterisk-1.4.18.1.tar.gz (18-Mar-2008 12:24 11M ) ?
Should be?
At the moment, it is.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL
Hello All,
Can anybody suggest bluetooth head phone which can be used to place calls
with eyebeam or any other soft phone.
Regards,
--
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email: [EMAIL PROTECTED]
MSN:
On 3/17/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Alex Balashov wrote:
Hello List,
I'm using a dialstring like the one below. I want to have three
different things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(2[:5000][:5000]))
These 3 things could
On Wed, Mar 19, 2008 at 1:32 AM, John Faubion [EMAIL PROTECTED] wrote:
when you look at the iPhone with all its amazing features for less than
$500.00 it just doesn't make sense. Am I the only one that thinks this?
Remember that the service providers such as ATT, Cingular, Sprint, Verizon
On Wed, 19 Mar 2008, David Quinton wrote:
On Tue, 18 Mar 2008 14:50:52 + (GMT), Gordon Henderson
[EMAIL PROTECTED] wrote:
BT nearly always try to sell featureline on business lines these days.
Would sir like a 3 of 5 year feature line contract?
Fair point, Gordon.
But in their defence,
YOu said '
(The only issue we've had with the TE121 is echo on voice calls, even
with the hardware echo cancelling module and lots of zapata.conf tuning...
did You EVER get the echo resolved ? How ?
Kevin DeGraaf wrote:
We used it in our installation and had some issues. We were passing
You Said I would probably go with an HP DL380.
would you share what drew you to that particular box ?
When i looked at their WEB page there was something like 9 different
RAID Controllers you could chose from.
Seemed hideously confusing to con fig one just to get a price quote on it.
Please
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to feed asterisk,
hoping to copy it to the outgoing directory all at 1 time.
___
-- Bandwidth and Colocation Provided by
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to feed asterisk,
hoping to copy it to the outgoing directory all at 1 time.
___
-- Bandwidth and Colocation Provided by
hi, all
I have configure tdm400p analog fxo card.
that's ok.
but how to chek that is working properly or not.
i chek with ztcfg - and zttool .
that's ok.
i want to dial from my fxo port to another extesion.
zaptel.conf
--
fxsls=1,2,3,4
defaultzone=in
loadzone=in
zapata.conf
I think the only way is managing the number of files in the outbound
directory
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Tong wrote:
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to
Tong wrote:
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to feed asterisk,
hoping to copy it to the outgoing directory all at 1 time.
Yes,
Create them with a future time and date and they will only be
On Wed, Mar 19, 2008 at 05:19:23PM +0530, Bhrugu Mehta wrote:
hi, all
I have configure tdm400p analog fxo card.
that's ok.
but how to chek that is working properly or not.
i chek with ztcfg - and zttool .
that's ok.
i want to dial from my fxo port to another extesion.
zaptel.conf
That might work.
I'll give that a shot.
Doug Lytle [EMAIL PROTECTED] wrote:
Tong wrote:
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to feed asterisk,
hoping to copy it to the outgoing directory
Hi all
I am playing with meetme and I have things working in speak live mode.
What I would like to do is have a small meetme that kicks off on a
schedule (no problem there)
and just plays a wave file in the meetme and then exists.
I have played with option b of the meetme where the AGI plays
On Tuesday 18 March 2008 22:12, Steve Totaro wrote:
For your use, I would go for a RAID 5
I would highly recommend against a raid 5 set. I can give you more details if
you are interested, but these guys have most if it down : www.baarf.com see
the link on the left on why should I not use Raid
On Tue, 2008-03-18 at 01:01 +0800, Pete Kay wrote:
It may seems like my lack of audio problem with PlayBack is due to
zaptel setting.
Yes, you're on the right track here. I'd be willing to bet dollars to
donuts you have a T1/E1 card that's not taking interrupts here. Get
your zaptel.conf
Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in
the voicemail messaging emailing operation. I had set serveremail option
to:
[EMAIL PROTECTED]
and under ast 1.2.x messages arrived at user mailboxes from
[EMAIL PROTECTED] . However, since upgrading emails
For more info, I grab the relevant portion of the maillog. It looks like
asterisk is trying to send using the right from email, but it's getting
changed. This would suggest a sendmail problem, EXCEPT, it works fine when
I send mail from the command line. Can anyone offer ideas?
Mar 19
Anciso, Roy wrote:
I ’ m trying to understand something that just doesn’t seem to
compute. How can companies like Cisco justify selling their hard
phones for as much as they do? I know there is a matter of recouping
RD costs but when you look at the iPhone with all its amazing
features
Yeah, I came accross that post too I think :) but as above, I already
tried moving the cables round, but no change.
Does anyone have a simular setup and can confirm theirs is fine?
4 Lines
1 x ADSL and Fax
3 x Voice
1 x Voice fails to answer call or dial out (no CID detected either)
2 x
what's a good way to apply a future date to thousands of files?
Doug Lytle [EMAIL PROTECTED] wrote:
Tong wrote:
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to feed asterisk,
hoping to copy it to
what's a good way to apply a future date to thousands of files?
Doug Lytle [EMAIL PROTECTED] wrote:
Tong wrote:
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to feed asterisk,
hoping to copy it to
what's a good way to apply a future date to thousands of files?
Doug Lytle [EMAIL PROTECTED] wrote:
Tong wrote:
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to feed asterisk,
hoping to copy it to
On Wednesday 19 March 2008 07:15:34 Doug Lytle wrote:
Tong wrote:
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to feed asterisk,
hoping to copy it to the outgoing directory all at 1 time.
Create them with
Tong wrote:
what's a good way to apply a future date to thousands of files?
I do it within the script that creates the file(s):
# Touch file once to set date/time stamp to now
/bin/touch /usr/local/bin/$1.out.call
# Touch file again with a 150 second future date
/bin/touch -r
And I can post a link that shows a bunch of guys think the earth is
flat with a 5/10 google ranking also (like the barf guys).
http://www.alaska.net/~clund/e_djublonskopf/Flatearthsociety.htm
I usually just call my guy at CDW and give him my needs, he is a
former techie gone sales. He puts
Getting closerthis seems to be a sendmail issue not asterisk.
It seems that if sendmail is run with -f (from) using any account other than
root, the from domain is NOT trusted, and so sendmail does ns lookup - which
of course resolves back to our firewall.
So it seems that I need to
On Wed, Mar 19, 2008 at 10:33 AM, Doug Lytle [EMAIL PROTECTED] wrote:
Tong wrote:
what's a good way to apply a future date to thousands of files?
I do it within the script that creates the file(s):
# Touch file once to set date/time stamp to now
/bin/touch /usr/local/bin/$1.out.call
On Wednesday 19 March 2008 10:36, Steve Totaro wrote:
And I can post a link that shows a bunch of guys think the earth is
flat with a 5/10 google ranking also (like the barf guys).
http://www.alaska.net/~clund/e_djublonskopf/Flatearthsociety.htm
Steve,
My purpose was to try to point out that
Gordon Henderson wrote:
On Sun, 25 Mar 2007, Gordon Henderson wrote:
and change the Optional Rule to:
3,3,7,1,0;10,4,7,2,0;60
someone correct me if I've goofed!
Well, it's been a year since I wrote this, and of-course I goofed and
no-one corrected me ;-)
Bother.
So I've
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,...and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,...and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my
Date: Wed, 19 Mar 2008 11:31:57 +0200
From: Atis Lezdins [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Handling 3 different call ending causes
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type:
I understand the maximizing pricing and branding aspect of phones but
when you look at feature set it just doesn't make sense. And as far as
purchasing the phone you can get it without a contract at the same
price.
When I starting thinking about it, can anyone else see a time when desk
phones
Mian M Asif wrote:
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,...and these user can call to each
other. Now i want to configure separate voice mail box for
Tong wrote:
That might work.
I'll give that a shot.
Doug Lytle [EMAIL PROTECTED] wrote:
Tong wrote:
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to feed asterisk,
hoping to copy it to the
RESOLVED! For others fiting a similar problem look at
/etc/mail/service.switch
This is the only way to force sendmail to not do a DNS lookup (first)...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: March-19-08 9:57 AM
To: Asterisk Users
Having ventured high enough and far enough to view the curvature of the
Earth and having stayed up late enough long enough (why do disks only
fail at the weekend?) to rebuild and restore RAID 5 sets, I proffer the
following (not so) Humble Opinion .
Dual power supplies, two thumbs up
but
Hi,
I am working on setting up the voice mail. I can get message recored and
can find the .wav file created. However, when I tried to play back, I can't
hear anything. In the CLI, it does say:
-- SIP/2000-081ed640 Playing
'/var/spool/asterisk/voicemail/default/2000/Old/msg' (language
RAID arguments (preference really) aside, 4k - 6k worth of student
voicemails is going to require quite a bit of storage space.
Thanks,
Steve Totaro
On Wed, Mar 19, 2008 at 12:01 PM, Drew Gibson [EMAIL PROTECTED] wrote:
Having ventured high enough and far enough to view the curvature of the
On Wed, 19 Mar 2008 11:32:44 -0400, Anciso, Roy [EMAIL PROTECTED] wrote:
When I starting thinking about it, can anyone else see a time when desk
phones are replaced by smart phones? Why would a company pay for work
cell phone and desk phone when one device could potentially do it all?
Our office averages around 1.5MB / mailbox, call it 10MB for rounding.
6,000 x 10MB = 60GB (n'est pas?)
2 x 250GB drives, mirrored, should cover that and the system quite nicely.
regards,
Drew
Disclaimer: Most of our employees are programmers so probably don't have
any friends to call and
This is not a troll. I've used my real email because I want this
taken seriously. I'm not trying to make anyone mad, I just want
some real discussion on this issue. Please bare with me...
I'm a USER of Asterisk. We purchased 3 commercially available
Asterisk Based PBXs a little over a year
Nice Ribbit Demo
http://blip.tv/file/753401
I think we should get some Asterisk video demo's up on blip.tv as well.
Post to this list with the url once you have your demo's up there.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney
On Wed, 19 Mar 2008, Paul Goodyear wrote:
Yeah, I came accross that post too I think :) but as above, I already
tried moving the cables round, but no change.
Does anyone have a simular setup and can confirm theirs is fine?
4 Lines
1 x ADSL and Fax
3 x Voice
1 x Voice fails to answer
What's wrong with YouTube? There seems to be quite a bit of content
there already
http://www.youtube.com/results?search_query=asterisk+pbxsearch_type=
Thanks,
Steve Totaro
On Wed, Mar 19, 2008 at 12:46 PM, Dean Collins [EMAIL PROTECTED] wrote:
Nice Ribbit Demo
I use standard wav (most compatible with players) so about a meg a minute.
In my experience, most people (users) use their voicemail similar to
email, they keep everything. Especially love struck college kids. I
think Asterisk has a soft limit of 1,000 (maybe it is 999) messages as
the max per
Bill Andersen wrote:
This is not a troll. I've used my real email because I want this
taken seriously. I'm not trying to make anyone mad, I just want
some real discussion on this issue. Please bare with me...
I'm a USER of Asterisk. We purchased 3 commercially available
Asterisk Based
Hello,
Thanks for all the tips.
But I'm totally lost at asterisk's world lol
For example, which property do I have to set to make a call center operator
busy?
How can I get the next client in the queue?
Can I have just one queue for web chat and phone calls? Can it be
integrated?
And about
Cool - nothing wrong with you tube - that's what I use for my corporate
publishing stuff.
Doesn't seem to be a lot of demo's like the ribbit demo though.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
-Original
On Mar 19, 2008, at 12:16 PM, [EMAIL PROTECTED]
wrote:
I understand the maximizing pricing and branding aspect of phones but
when you look at feature set it just doesn't make sense. And as
far as
purchasing the phone you can get it without a contract at the same
price.
When I starting
John Novack wrote:
Bill Andersen wrote:
This is not a troll. I've used my real email because I want this
taken seriously. I'm not trying to make anyone mad, I just want
some real discussion on this issue. Please bare with me...
I'm a USER of Asterisk. We purchased 3 commercially
Hello,
How do I install phpagi?
http://phpagi.sourceforge.net/
I couldn't find any info about setup in that site, and I couldn't email the
developers.so I'm lost.
I know it isn't a real question for this list, but I suppose many people
here already have installed it.
So, how can
On Wed, 19 Mar 2008, Norman Franke wrote:
As for why a company would purchase hard phones, several reasons. First, we
are replacing many hard phones with computers. We have a custom application
and have been moving folks main numbers to use the computer. We can make it
ring externally and
You don't install it as such, you just include the files from your php
scripts.
On 19/03/2008, Carlos Carvalhar [EMAIL PROTECTED] wrote:
Hello,
How do I install phpagi?
http://phpagi.sourceforge.net/
I couldn't find any info about setup in that site, and I couldn't email
the
On Wed, Mar 19, 2008 at 12:48 PM, Carlos Carvalhar
[EMAIL PROTECTED] wrote:
How do I install phpagi?
http://phpagi.sourceforge.net/
Since phpagi is really just a set of php libraries, all you need to do
to install is dump it somewhere and add that location to your php
include_path.
-Erik
I am trying to use meetme() on SIP channels.
I found this line on voip-info.org
-
It *is* necessary either to have a Digium card or a dummy timing driver
(e.g. ztdummy or zaprtc) in order for MeetMe to work at all, but that
doesn't help you use AGI with SIP channels: They have no
On March 19, 2008 12:43:21 pm Bill Andersen wrote:
I'm a USER of Asterisk. We purchased 3 commercially available
Asterisk Based PBXs a little over a year ago. (I won't mention
which one at this point - I don't want to bad mouth them - yet!)
Two of the systems are very small (5 SIP lines/6
how many out going files were you working with?
Steve Totaro [EMAIL PROTECTED] wrote:
On Wed, Mar 19, 2008 at 10:33 AM, Doug Lytle [EMAIL PROTECTED] wrote:
Tong wrote:
what's a good way to apply a future date to thousands of files?
I do it within the script that creates the
On Wed, Mar 19, 2008 at 11:43:21AM -0500, Bill Andersen wrote:
This is not a troll. I've used my real email because I want this
taken seriously. I'm not trying to make anyone mad, I just want
some real discussion on this issue. Please bare with me...
2) Are there any users out there that
yeah,
I saw at their website a simple php include:
require_once('../phpagi-asmanager.php');
But when I download the gz file it doesn't uncompress as php files, the
phpagi-2.14.gz file returns a phpagi-2.14 file...and I tried with winrar and
7-zip that usually uncompress gzip files without
John
You have raised few valid points. Thanks.
However, I will say that it is not asterisk but people/company
deploying
it. Generally speaking after deployment, and as long users are using
the system normally, no reboot is required.
And yes, running the whole thing from standard PC
Bill Andersen wrote:
This is not a troll. I've used my real email because I want this
taken seriously. I'm not trying to make anyone mad, I just want
some real discussion on this issue. Please bare with me...
I'm a USER of Asterisk. We purchased 3 commercially available
Asterisk Based
Ya i had some months ago. it works fine. what you need to know else...
JehanZaib Younis
Date: Wed, 19 Mar 2008 14:04:35 +0500From: [EMAIL PROTECTED]:
asterisk-users@lists.digium.com; [EMAIL PROTECTED]: Query about Bluetooth Head
phone
Hello All,
Can anybody suggest bluetooth head phone
On Wed, Mar 19, 2008 at 1:31 PM, Carlos Carvalhar
[EMAIL PROTECTED] wrote:
But when I download the gz file it doesn't uncompress as php files, the
phpagi-2.14.gz file returns a phpagi-2.14 file...and I tried with winrar and
7-zip that usually uncompress gzip files without problem.
How
On Wed, 19 Mar 2008, Senad Jordanovic wrote:
And yes, running the whole thing from standard PC based desktop will
eventually cause issues hence an solid state appliance is a way to go :)
My gripe is that I think people try to put too much into a system, don't
have a server build and
130 physical extensions including 24x7 inbound call centre
Debian on Dell server
[EMAIL PROTECTED]:~# uptime
13:15:31 up 192 days, 23:49, 2 users, load average: 0.00, 0.01, 0.00
here is one more running multi tenant Hosted PBXes:
saul ~ # uptime
18:59:11 up 263 days, 23:50, 1
I've been searching to a solution to this for a while and can't
figure it out, perhaps someone has done something similar.
I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to
my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low
on my lightly loaded switched
On Wed, Mar 19, 2008 at 06:54:46PM +, Gordon Henderson wrote:
On Wed, 19 Mar 2008, Senad Jordanovic wrote:
And yes, running the whole thing from standard PC based desktop will
eventually cause issues hence an solid state appliance is a way to go :)
My gripe is that I think people
An off-the-shelf 5+ year old MSI MS-6378X-L motherboard, 1.6GHz AMD, 512
RAM, 10 extensions, no more than three concurrent calls:
[EMAIL PROTECTED] ~]$ uptime
11:31:45 up 103 days, 1:00, 2 users, load average: 0.00, 0.00, 0.00
But:
[EMAIL PROTECTED] ~]$ sudo asterisk -rx 'core show uptime'
On Mar 19, 2008, at 2:48 PM, [EMAIL PROTECTED]
wrote:
My mobile does not sound terrible, does not have echo, does not
fade in or
out, and the last time I used it to call the emergency services, I got
through straight away. I've not had a dropped call for a long time
either
(going through
On Mar 19, 2008, at 1:00 PM, [EMAIL PROTECTED]
wrote:
Am I expecting too much?
Perhaps.
I think the hardware on which we run Asterisk can be much more
reliable than the software, which is often the case. We have a bunch
of HP servers with RAID and have never lost anything. A HD may
Senad Jordanovic wrote:
130 physical extensions including 24x7 inbound call centre
Debian on Dell server
[EMAIL PROTECTED]:~# uptime
13:15:31 up 192 days, 23:49, 2 users, load average: 0.00, 0.01, 0.00
here is one more running multi tenant Hosted PBXes:
saul ~ # uptime
18:59:11 up 263
Tong wrote:
how many out going files were you working with?
Me or Steve? I've never done anything more then 5.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
How would I setup chanspy from a call file???
I want to make a call file that uses a Local channel and plays a wave file.
Then I want to make a call file that calls an extension
and does a chanspy on the local channel above that is playing a wave file.
How do I set that up?
Jerry
On Wed, 19 Mar 2008, Tzafrir Cohen wrote:
On Wed, Mar 19, 2008 at 06:54:46PM +, Gordon Henderson wrote:
On Wed, 19 Mar 2008, Senad Jordanovic wrote:
And yes, running the whole thing from standard PC based desktop will
eventually cause issues hence an solid state appliance is a way to go
Senad Jordanovic wrote:
However, I will say that it is not asterisk but people/company
deploying it. Generally speaking after deployment, and as long
users are using the system normally, no reboot is required.
I'm thinking part of the problem IS the company deploying
the commercial product we
Bill Andersen wrote:
a) IF... I expect a phone system to just work. Once it is
configured, a phone system should just work with
very little attention. My previous system was a
Comdial with external voice mail on a DOS based PC.
I LITERALLY WENT OVER 4 YEARS
Finally got my Cisco Call Manager link going; what it turned out to be was
having the same extension on the Asterisk system and on the Call Manager
side of things. Changing the extension on one side fixed it. Which brings me
to...
I need to have the same extensions on two sites. So if I use an
Andrew Kohlsmith (lists) wrote:
If you're continuously restarting Asterisk, there is something wrong
with your setup: hardware, software or both. I have many installs
out there on commodity hardware (either pure-voip or digital (PRI)
only with Polycom handsets) and none of them need to be
The box has been up since we upgraded the UPS, time before was for the
disk failure in Feb 2007.
Asterisk has now been up for 5 hours, 44 minutes (yes, by Murphy's Law,
I'm troubleshooting a problem butrestart when convenient does not
impact real uptime) but yesterday it had been up for 63+
Drew Gibson wrote:
The box has been up since we upgraded the UPS, time before was for the
disk failure in Feb 2007.
Asterisk has now been up for 5 hours, 44 minutes (yes, by Murphy's Law,
I'm troubleshooting a problem butrestart when convenient does not
impact real uptime) but yesterday
Thank you to everyone that replied to my post. I started to
reply to most of them, but it is getting a little out of hand.
Again, thank you. It actually makes me think the problem is not
so much with Asterisk as it is with implementation. (My Vendor)
Although this is a users list, I think it is
Bill Andersen wrote:
Senad Jordanovic wrote:
However, I will say that it is not asterisk but people/company
deploying it. Generally speaking after deployment, and as long
users are using the system normally, no reboot is required.
I'm thinking part of the problem IS the company deploying
On Wed, Mar 19, 2008 at 4:38 PM, Bill Andersen [EMAIL PROTECTED] wrote:
Although this is a users list, I think it is more of a list
for Asterisk resellers. I'd be interested in how many of you
are simply using Asterisk as your phone system and NOT selling
your services or an Asterisk
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