Hello,
I notice that at the end of the day, after about 4000 calls have passed
my Asterisk-system, that the use of memory is very high and stays that
way untill a restart of Asterisk or a reboot of the server.
This is the situation at the end of the day :
[root@sp asterisk]# free -m
Hi there
How can I reset the value of asterisk' calls processed without restarting
asterisk? Where does it save/access the value of all processed calls since
last restart from?
--
_
-- Bandwidth and Colocation Provided by
AFAIK:
Linux has a tendency to keep RAM filled up with any recently accessed
progarm. To keep programs access fast enough, it never removes something
from the memory, only replaces it, in case some program has more frequent
access than the one already present in ram.
If your server isn't
Nope, no swapping...
Thanks.
Jonas.
On 03/08/2012 09:44 AM, [Digital^Dude] ® wrote:
AFAIK:
Linux has a tendency to keep RAM filled up with any recently accessed
progarm. To keep programs access fast enough, it never removes
something from the memory, only replaces it, in case some program
Is this a general issue or just affecting specific versions?
Jon Farmer
Tel 07795 118140
On 7 March 2012 16:33, Paul Belanger pabelan...@digium.com wrote:
On 12-03-07 04:29 AM, Jon Farmer wrote:
Hi
I have recently upgraded a box to 1.8.9.3 and have noticed that
randomly the logger will
Hi
Just realised this is due to a FIFO blocking. Fixed that and all back to normal.
Regards
Jon
Jon Farmer
Tel 07795 118140
On 7 March 2012 16:33, Paul Belanger pabelan...@digium.com wrote:
On 12-03-07 04:29 AM, Jon Farmer wrote:
Hi
I have recently upgraded a box to 1.8.9.3 and have
Le 07/03/2012 11:21, Administrator TOOTAI a écrit :
Le 07/03/2012 09:46, Markus a écrit :
Am 07.03.2012 02:04, schrieb Mike Diehl:
I tried the chat as well with no effect. My German is a bit rusty,
or I'd call
them
Most Germans speak English. :)
Well, I sended them email on
Hi
I set the debug to 15, and changed to peer, I got this:
--- SIP read from UDP:94.77.210.xxx:5060 ---
SIP/2.0 500 account has been moved to a remote system
Via:
Hi Bilal
in my case i use an IVR menu using asterisk 1.4 an i can store the number
of the customer in my database and after i can select
the phone number and the date_time of calling i use mysql
you must change database login password with yours and also the name of
table
regards
exten =
I need call to C every time that A call to B, but when A-B hangs up i need
to hang up Asterisk-C call too.
Anyboby know another solution?
On Wed, Mar 7, 2012 at 2:51 PM, equis software equissoftw...@gmail.comwrote:
Here's my dialplan...
[default]
exten = _X.,1,System(echo -e Channel:
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
What you want to do is complicated with Asterisk. Your best solution may be to
write an application to monitor active calls via the Asterisk Manager interface.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
I had submitted a patch some time ago to add option s to chanspy. This would
cause chanspy to exit once the specified change was not longer there. I do not
know if it ever got into a released version as I use ABE. It was not in 1.6 but
might be in 1.8.
--
Jim Dickenson
Hi all,
We're testing TLS and SRTP on Asterisk 1.8.10.0 and have it working
with a commerical (not self-sign) AlphaSSL wildcard (GlobalSign) using
Blink Lite 1.6.2 as per
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
We've tested with Bria on an iPhone and that doesn't
AFAIK, this is a shell count (The count is kept in shell memory for the
running asterisk process). You handicap potential answer by not stating
your Asterisk version or your technology (SIP/DAHDI/T1/etc). If you are
just using SIP trunks, SIP RELOAD might do it.
From:
On 03/08/2012 09:32 AM, Gavin Henry wrote:
Hi all,
We're testing TLS and SRTP on Asterisk 1.8.10.0 and have it working
with a commerical (not self-sign) AlphaSSL wildcard (GlobalSign) using
Blink Lite 1.6.2 as per
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
We've tested
Our experience has always been, with all versions of Asterisk, that when
you do an originate command from Asterisk CLI then it stops showing CLI
verbose and one has to open a new terminal to see the verbose. Logging
status is in disarray in all version. These behaviours breaks security
tools
On Wed, 7 Mar 2012, bilal ghayyad wrote:
If I need to build IVR using Asterisk (so I will read and write to
database), until now from my reading, I can understand that the best way
is to use AGI to call external script like php which will manipulate
every thing, correct?
I have a strong
Un top-posting...
On Thu, Mar 08, 2012 at 09:47:54AM +0100, Jonas Kellens wrote:
On 03/08/2012 09:44 AM, [Digital^Dude] ® wrote:
AFAIK:
Linux has a tendency to keep RAM filled up with any recently accessed
progarm. To keep programs access fast enough, it never removes something
from the
On 12-03-08 04:41 AM, Jon Farmer wrote:
Hi
Just realised this is due to a FIFO blocking. Fixed that and all back to normal.
How did you fix it? Will help others playing at home (and me too).
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Apologies for the top post, something is screwed up with my email client,
will fix it soon.
What a BS story that I have debunked many times. A used Key System could
be purchased for a few hundred dollars, a much better investment then
writing your own PBX from scratch.
A company that is
My question is someone (Digium) must have this working against Polycom
(which is a requirement for this project) with commercial certs since
that's their partner of choice?
I don't believe we've done any interop testing with Polycom phones since TLS
and SRTP support were added to Asterisk.
On Thu, 8 Mar 2012, [Digital^Dude] ® wrote:
How can I reset the value of asterisk' calls processed without
restarting asterisk? Where does it save/access the value of all
processed calls since last restart from?
(I'm just a 1.2 Luddite, so my input may be a bit dated.)
Where are you seeing
1.4:
pbx core show channels
[snip]
167 active channels
84 active calls
1.8:
pbx core show channels
[snip]
23 active channels
12 active calls
9567 calls processed
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
On 03/08/2012 10:34 AM, Gavin Henry wrote:
My question is someone (Digium) must have this working against Polycom
(which is a requirement for this project) with commercial certs since
that's their partner of choice?
I don't believe we've done any interop testing with Polycom phones since TLS
Ah, this makes sense now. So as of today the status of TLS and SRTP in
anything
other than 1.4.X is unknown?
Umm... no :-)
OK, sorry :-)
Asterisk 1.4 did not have support for SRTP or SIP/TLS. Thus, neither of
these were tested with Polycom phones the last time we did interop testing
AFAIK, it works in the 1.8 and 10.X branches (I have used it in 10.0.2)
There was a known issue with some certificates that used multiple levels
IIRC.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gavin Henry
http://adhearsion.com/ is cool if you are familiar with ruby programming
language.
On Thu, Mar 8, 2012 at 9:55 PM, Steve Edwards asterisk@sedwards.comwrote:
On Wed, 7 Mar 2012, bilal ghayyad wrote:
If I need to build IVR using Asterisk (so I will read and write to
database), until now
Same question for asterisk-users as well:
- Forwarded message from Tzafrir Cohen tzafrir.co...@xorcom.com -
Date: Wed, 7 Mar 2012 21:14:04 +0200
From: Tzafrir Cohen tzafrir.co...@xorcom.com
To: asterisk-...@lists.digium.com
Short version: it's now time to remove.
Anybody actually uses
Thank you Eric for helping me out. I am using asterisk 1.6.x, 1.8.x
On Thu, Mar 8, 2012 at 9:48 PM, Eric Wieling ewiel...@nyigc.com wrote:
1.4:
pbx core show channels
[snip]
167 active channels
84 active calls
1.8:
pbx core show channels
[snip]
23 active channels
12 active calls
On Thu, 2012-03-08 at 16:50 +, Gavin Henry wrote:
Ah, this makes sense now. So as of today the status of TLS and SRTP in
anything
other than 1.4.X is unknown?
Umm... no :-)
OK, sorry :-)
Asterisk 1.4 did not have support for SRTP or SIP/TLS. Thus, neither of
these were
Hi all,
I've done a basic install of 10.1.2 to have a play with the new
ConfBridge application and have noticed high latency when in a
conference. It's to the order of 900ms or so which is just too much for
a conference to work well.
I can account for about 120ms of that latency, but not
Hi All;
Really I need to know why when using the h in the exten =, then we use
DeaAGI with it?
I am using vicidial and I see this line alot, so I need to know how it work
(when it will be executed):
exten =
On Thu, 8 Mar 2012, bilal ghayyad wrote:
Really I need to know why when using the h in the exten =, then we
use DeaAGI with it?
This is weird. I entered 'deadagi' in Google's search box and the first
link was right on topic!
When this line will be executed? After the channel will be hanged
Danny,
I use 1.6.x and 1.8.x asterisk versions. I would think asterisk call
counters won't be changed in each version... hence I thought the asterisk
version wouldn't be relevant. Reload of a particular application doesn't
reset the counters. I have noticed that calls done with AMI Originate,
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