On Thu, May 12, 2011 at 09:44:31PM -0400, Jose P. Espinal wrote:
Hello Folks,
What could be producing the following warnings on console, after an
installation from source (Asterisk 1.4.41):
[May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module:
Error loading module
I was looking for MySQL table structures for ARA (Asterisk 1.8.X).
I found one for SIP friends on,
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure
But it seems that it is not as per the Asterisk 1.8 SIP options. i.e. it
contains 'call-limit' which is deprecated in
Probably using XML - which is phone dependant.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 12 May 2011 21:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Cor-wrong (sort of).
There is a backport of DevState/Device_State for 1.4
https://issues.asterisk.org/view.php?id=15818
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
Wieling
Sent: 12 May 2011 20:01
Andrew Thomas wrote:
Cor-wrong (sort of).
There is a backport of DevState/Device_State for 1.4
https://issues.asterisk.org/view.php?id=15818
Very cool! I'll have to review this weekend.
Thank you!
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
Im running v1.8.2.3 and not have no had this issue you speak of?
I saw it once or twice, but otherwise, it works.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jeremy Kister
Sent: Thursday, May 12,
Glad you solved it. Now I'm having high CPU load issue. I don't know
why but sometime my asterisk process reached ~150% CPU load and just
locked no calls nothing only solution is kill -9
I've 1000hz preemtive kerenel on ubuntu do you think it's the issue
because of low through put ?? Which
Hi,
Sometimes calls on Asterisk fail to connect to DAHDI channels and giving
below error:
Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel
congestion)
There are 8 E1 connected on server and only 15-20 simultaneous calls. All
channels and E1 are showing in service without
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
deeps backup
Sent: Friday, May 13, 2011 9:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DAHDI Error
Hi,
Sometimes calls on
Tzafrir Cohen wrote:
[May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module:
Error loading module 'res_musiconhold.so':
/usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol:
cap_set_proc
Could this be related to having used 'strip' on the binaries?
Note: I have
On 13 May 2011 14:06, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
deeps backup
Sent: Friday, May 13, 2011 9:02 AM
To: asterisk-users@lists.digium.com
hi:
I am using 64bit scientific linux 6 with default kernel. my
loading is quite low, maybe 1~10 concurrent calls. I remember last
time I have unstable problem about timer.
my linux now use HPET clock. and asterisk use res_timing_dahdi instead
of the default res_timing_timerfd. I don't know if
Hi Guys:
I am very new in Asterisk Queue, so may be i'm doing wrong somewhere.
I have Asterisk 1.8.3.3 and Dahdi 2.4.1.2.
I defined some agent's on Asterisk Queue, and the problem is that the agent
is allways on UNKNOWN status, so Asterisk can dial to the agent even if the
agent is allready
Hi,
Here http://www.voip-info.org/wiki/view/Asterisk+func+device_State you can
find a link to download a backported for Asterisk 1.4 version of
DEVICE_STATE function.
(Elsewhere, you can find reference to another backported function DEVSTATE
which seems to behave the same as DEVICE_STATE).
As I
Hi All,
I'm using Asterisk 1.8.2 with outbound calls using Google Voice. I've been
having issues calling several toll free numbers where the call 'is ringing'
but never transitions to 'answered'. These are toll free numbers which are
typically answered by an ivrs where you enter eg. a conference
Thanks for reply,
How do i find asterisk using which timing res_timing_timerfd or
res_timing_dahdi ?
-S
Date: Fri, 13 May 2011 22:13:47 +0800
Subject: Re: [asterisk-users] 1.8 and prematuremedia problem
From: tbs...@gmail.com
To: satish...@hotmail.com; asterisk-users@lists.digium.com
I didn't understand very well.. So you cant dial on the first 24 channels?
Did you take care on the jumper of the card?. There is something related to
E1 (31 channels) or T1 (24 channels).
And check the system.conf either.
rv
2011/5/13 deeps backup backup.de...@gmail.com
I have checked
Show us a pri debug of a problem call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
deeps backup
Sent: Friday, May 13, 2011 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sorry for top post I'm responding from my blackberry
I haven't tried with timerfd but with timer pthread 1.8 is very unstable
I think I have seen a post to the list from kevin fleming that the same is for
timerfd that there is a nasty bug which they haven't found the reason for yet
+1
/@
\ \
___ \
(__O) \
(@) \
(@) \
(__o)_\
\\
On Tue, May 10, 2011 at 4:23 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
I'll keep this brief because I don't want to come across like any more of an
a$$ than I absolutely have to, especially
I can dial 1-24 channels but not after that. There are 8 E1s. Box was
working fine and carrying traffic on all E1s before. Just recently i noticed
this problem has occurred.
On 13 May 2011 16:30, Rafael Visser visser.raf...@gmail.com wrote:
I didn't understand very well.. So you cant dial on
sangoma cards do not use dahdi...
13.5.2011 v 17:16, satish patel satish...@hotmail.com:
Thank you so much!! I found following (res_timing_timerfd.so in USE). But we
have asterisk dahdi install and sangoma A102D pri card configured. Do you
think i should use res_timing_dahdi.so ?
You mean say i don't use res_timing_dahdi.so ? I guess this is just timing
module nothing related to Card.
_S
From: tu...@canistec.com
Date: Fri, 13 May 2011 18:30:52 +0200
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 1.8 and prematuremedia problem
sangoma cards do not
On Thu, 2011-05-12 at 22:17 +0200, Jonas Kellens wrote:
On 05/12/2011 07:12 PM, Carlos Chavez wrote:
On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote:
Hello,
is there some way to make Asterisk light up a certain light on an
IP-phone ?
Like MWI, the message waiting
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom
On Fri, May 13, 2011 at 11:07 PM, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6:
On 5/13/11 10:57 AM, RSCL Mumbai wrote:
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like
Hi,
Needed to test follow-me this evening on Asterisk 1.6.2.17 and received the
following message:
== Spawn extension (international-US, 0114407590XX, 5) exited non-zero on
'Local /0114407590XX@aXX-a62a;2'
-- no live channels left. exiting.
I have not seen that before. What does
Hi all,
Anyone know how to make asterisk properly reply to options keep-alive? Or
just force a 200 OK somehow?
I recently took over a server and they have ~80 pap2 devices that send nat
keep-alive and * always replies with 481 No subscription. It's more of an
annoyance, I know but I
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler
Sent: Friday, May 13, 2011 2:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] OPTIONS Keep alive -
On Fri, May 13, 2011 at 2:58 PM, Skyler skchopper...@gmail.com wrote:
Hi all,
Anyone know how to make asterisk properly reply to options keep-alive? Or
just force a 200 OK somehow?
I recently took over a server and they have ~80 pap2 devices that send nat
keep-alive and * always
Really!? Wow, that would be so easy as it looks like qualify=yes is already
enabled on each SIP channel. I'll give that a try/test first and report
back.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Friday,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler
Sent: Friday, May 13, 2011 3:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] OPTIONS Keep alive -
On 05/12/2011 02:46 PM, Jason Parker wrote:
I'll make it a point to respond to this email when the new builds are available.
These builds are now available.
--
_
-- Bandwidth and Colocation Provided by
On Thu, 2011-05-12 at 21:30 +0200, bakko wrote:
Hi,
look if you have res_config_mysql.so module instaled on your asterisk.
On CentOS /usr/lib/asterisk/modules
Regards
Tnx for your reply.
It turned out, that mysql-support was in a different rpm (addons)
As systems are never connected to
BTW, is GTalk/Jabber a part of RPM now?
-Vladimir
On 5/13/2011 5:43 PM, Jason Parker wrote:
On 05/12/2011 02:46 PM, Jason Parker wrote:
I'll make it a point to respond to this email when the new builds are
available.
These builds are now available.
--
Hi,
On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest.
Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly
1-2 concurrent calls. No other activity on server. Top shows asterisk on
top.
Its quad xeon server with 4 gb ram.
Any suggestion where should I
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