Hi All,
We are currently evaluating the confbridge application while we prepare to
upgrade our environment to asterisk v1.6.2.x. We have run in to two issues
using it to kick/mute participants in a bridge and would like to ask for the
experience of others running the application for any
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On Wed, 2010-06-09 at 19:43 +, Edwin Quijada wrote:
Just is PRI line you can do it..
No, not so.
I both have some PRI and BRI lines.
All of them have a main-number, and some additional numbers
Depending on what contract you have with your ISDN-provider the amount
of those number can vary.
Hi all,
I think i understand the problem, actually I have two asterisk server. In
the extension.conf file on one server I have added
exten = 3923903,1,GOTO(s,1,3923903.conf)
which reads the corresponding conf file when ever the extension no. through
PSTN is called and learns the location of
Hi,
It isn't a problem with the list. And it is not 'mine'. It is a problem with
your software. I am just one of the thousands of people it is annoying! Perhaps
your IT staff could help fix it?
CCing the list so everyone is aware of your wonderful customer service. ;)
S
On 10 Jun 2010, at
Hi,
I am using Asterisknow 1.5. And TDM400P card for interfacing with PSTN
line. This setup was working without any problem. But now it is showing
issues. When I try to call through PSTN, there is a continuous large noise
is hearing from the SIP phone. And can't make the call. When I try to
Hello list,
is it possible to group some peers and limit their overall call
limit?
Ex: 4 peers can make max 2 concurrent calls.
Thanks in advance,
Alex
--
_
-- Bandwidth and Colocation
Shouldn't a moderator block emails from this email address, maybe
temporarily?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-10 6:41 AM, Steve Howes steve-li...@geekinter.net wrote:
Hi,
It isn't a problem with the list. And it is not 'mine'. It is a problem with
your software. I am just
I'm getting a lot of these on the CLI :
[Jun 10 13:41:36] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of
0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not
permitted
[Jun 10 13:41:37] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of
0x1b381bb0 (len 554) to
On 10/06/10 11:56, Alexandru Oniciuc wrote:
Hello list,
is it possible to group some peers and limit their
overall call limit?
Ex: 4 peers can make max 2 concurrent calls.
Thanks in advance,
Alex
Hi
you can use
call-limit
in the sip.conf at a peer
Hi list,
is there a way to achieve in asterisk (version 1.4.x) the behavior
described below?
* a caller place a call to an extension, and I want the caller hears
the extension ringing for some seconds, and then hears the music
on hold (or a courtesy message) _in the same call;_
Here is one way to do it (works in 1.4.22-1.4.30 at least)
exten = s,n,Dial(SIP/,10)
exten = s,n,Dial(SIP/,90,m(default))
This snippet will ring for 10 seconds with Ringing, then ring for
90 seconds or until answered with MOH.
_
From:
Ok Danny but with this example I have 2 calls in the called phone, and
this is what I have to avoid!
Regards,
Matteo
Il 10/06/2010 15.16, Danny Nicholas ha scritto:
Here is one
way to do it (works in
1.4.22-1.4.30 at least)
exten = s,n,Dial(SIP/,10)
exten =
Dear all
using top -H i can see that some asterisk thread are consuming many
CPU (sometimes more than 50%)
Is there a way to understand what is doing the process with pid 9429 ?
i've tried the core show thread command, but it doesn't seem to print
any PID information.
Thanks to all in advance
On Thu, 10 Jun 2010, Zeeshan Zakaria wrote:
Shouldn't a moderator block emails from this email address, maybe
temporarily?
There is no moderator.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com
On Thu, Jun 10, 2010 at 03:46:31PM +0200, nik600 wrote:
Dear all
using top -H i can see that some asterisk thread are consuming many
CPU (sometimes more than 50%)
Is there a way to understand what is doing the process with pid 9429 ?
strace -p 9429
This would help if the thread
Not sure how this would work, but you could create a special MOH file that
was 10 seconds of ringing followed by the normal MOH - I know this CAN be
done, just takes a bit of trial and error.
_
From: asterisk-users-boun...@lists.digium.com
If Doug generates enough spam with his unfortunate rule selection, he will
probably get zapped next month; We just have to live with it until the
14th.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwin,
In your outbound context, you need to have the dialplan evaluate the
hangupcause variable and send an appropriate message to your callers.
Check out the following URL for some samples that you may adapt for your
circumstance.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-10 10:19 AM, Danny Nicholas da...@debsinc.com wrote:
If Doug generates enough spam with his unfortunate rule selection, he will
probably get zapped next month; We just have to live with it until the
14th.
-Original Message-
From:
I remember at least once, may be two years ago, similar out of office
replies were flooding this mailing list almost once every hour or two, and
that email address was blocked, with a confirmation to the list that the
address was blocked.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-09
Hi Guys,
I have Spikko setup as provider of DID and outbound routes and I can make
calls out but no inbound calls via DID can be made. I did a sip debug which
is reported below. I never receive the call though, I have a catch all in my
inbound routes and it doesn't hit my context at all or not
I called Mary and chatted about how to suspend Doug's subscription to the
list.
She's doing her best to take care of it, so let's cut her a little slack :)
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
2010/6/9 Olivier oza_4...@yahoo.fr
Hi,
Is Astmanproxy still downloadable ?
At the moment, I can't download anything.
I'm usually using this
http://github.com/davetroy/astmanproxy/tarball/master URL
I can use a previous tar file but I would be pleased to know if I should do
something
On Wednesday 09 June 2010 16:28:44 nik600 wrote:
Reading the the upgrade file it seems that the pbx_realtime should
affect also the extension.conf settings... where am i wrong?
You're just wrong. Extensions.conf is not affected at all by that setting.
--
Tilghman Lesher
Digium, Inc. | Senior
FreePBX questions should be asked at FreePBX forums.
As for the asterisk part, where are you defining the context to receive
incoming calls? Probably in the trunk settings (Peer Details) you need to
add context=from-trunk if FreePBX still uses it as the default context for
incoming calls.
Thanks Don - Doug isn't going to be a happy camper when Mary gets done with
him...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Thursday, June 10, 2010 10:18 AM
To: 'Asterisk Users Mailing
Doug will have it easy.
I pity the next member of this group that forgets to take care of business
before going on vacation!
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent:
We have been distributing asterisk servers for several years now, and early on
decided that hardware echo can was the way to go. Our first few boxes without
it had horrid echo problems, and attempts at tuning in 2006 didn't make any
difference.
We installed a new server yesterday at a
At 09:49 AM 6/10/2010, you wrote:
The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand that
when Dahdi detects no HWEC, it enables SWEC by default. Is there
anything I can
do to tweak the settings to try and make this liveable for the
client until we
get the card? The server is in
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
We are totally out of touch on the subject of software echo cancellation in
asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand
that
when Dahdi detects no HWEC, it enables SWEC by default. Is there anything I
can
do to
- Gordon Henderson gordon+aster...@drogon.net escreveu:
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
We are totally out of touch on the subject of software echo
cancellation in
asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I
understand that
when Dahdi detects no
On Thu, 10 Jun 2010, Gordon Henderson wrote:
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
We are totally out of touch on the subject of software echo cancellation in
asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand
that
when Dahdi detects no HWEC, it enables SWEC
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
On Thu, 10 Jun 2010, Gordon Henderson wrote:
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
We are totally out of touch on the subject of software echo cancellation in
asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via sip-account.
My extension conf is:
general]
static=yes
writeprotect=no
[globals]
OUT_PORT=1
[ISDN]
exten =
I may not have made myself clear. I'm not actually trying to change
anything. I'm just trying to figure out what is happening. (I'm
trying to analyze log files as part of evaluating our call center's
performance.)
I do know we don't have weights set for our queues.
On 6/10/10, Mike
On Thu, Jun 10, 2010 at 07:25:43PM +0100, Gordon Henderson wrote:
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
Isn't OSLEC on by default?
Sadly it's not included in the default DAHDI. Several distros include
it. Those also set it as the default EC for system.conf generated by
dahdi_genconf.
Danny Nicholas wrote:
Not sure how this would work, but you could create a special MOH file
that was 10 seconds of ringing followed by the normal MOH – I know this
CAN be done, just takes a bit of trial and error.
That's what I would suggest as well. You could use Monitor() initially to call
On Fri, Jun 11, 2010 at 12:19:37AM +0200, Gergo Csibra wrote:
Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote:
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via
I'm trying to figure this out. I have agents answer calls from two
different queues. We have things set up so that these agents only
see one call at a time. Let's say an agent picks up a call while
there are calls waiting in both queues. Clearly the head of one of
the queues will now start
On Thu, 10 Jun 2010, Gordon Henderson wrote:
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
On Thu, 10 Jun 2010, Gordon Henderson wrote:
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
We are totally out of touch on the subject of software echo cancellation in
asterisk. The system is
I'm having similar errors too.
Customers are also complaining that their connection dropped. Could this be
related?
(That's the only error that occured when that connection dropped)
(using asterisk 1.6.2.8)
Kenny
--
tel: +32 476 780 692
On 10 Jun 2010, at 13:52, Jonas Kellens wrote:
I'm
I am having problems with Eyebeam when the user dials a number that is
not available. This problem exists with both Asterisk 1.4 and 1.6 using
Eyebeam or Xlite. The problem seems to be that when the soft phone
receives the 503 Unavailable response it will not be able to dial
another number
Hi,
Isnt there a parameter called weight for each queue that defines exactly
that? I never tried it, but it appeared to do exactly what you want
(according to the invaluable but often out-dated wiki).
Regards,
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi!
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via sip-account.
Unless you are using mISDN v2: Do yourself a favour and switch to CAPI
with chan_capi and fcpci. mISDN v1 is
Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote:
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via sip-account.
Unless you are using mISDN v2: Do yourself a favour and
I'm looking for a small formfactor mobo for an install that needs to handle 25
phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone
know what kinds of call volume that will handle?
MD
--
_
-- Bandwidth
Based on comments from Ward Mundy during a recent VUC call I'd expect
even a single CPU Atom system to handle that many phones in an office
application. Perhaps there may be merit in dual CPU in more of a call
center application.
I don't know how it calculates it, but FreePBX shows a bar for total
calls that looks like it maxes out at six. We haven't hit that on any
installs of this device yet, but that seems pretty low for sure.
I know with four calls in progress, all VoIP, transcoding G711u to G.729,
the load of
On Thu, 10 Jun 2010, Michelle Dupuis wrote:
I'm looking for a small formfactor mobo for an install that needs to
handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz
mobo - anyone know what kinds of call volume that will handle?
On Thu, 10 Jun 2010, mgra...@mstvp.com
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