Hi all,
I'm trying to tighten things up a bit and I seem be be running into something
that doesn't make sense to me.
I've got 2 contexts, one for customers, and one for guests, that I include
into [customers] and [default], in extensions.conf, as below:
I worked with Project Honeypot guys for a while, they are more than
willing to assist, as they already have the backend work done for a
clearing house identifying hackers. The biggest issue we had a year
ago was to create the mechanism in asterisk to push valid log messages
out to the
On Mon, Apr 12, 2010 at 7:17 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
That only addresses EC2 (and assumes that Amazon has any interest in
protecting their reputation). What about attacks that come from other
locations? Granted it's pretty easy to buy time on an EC2 server so
On Mon, Apr 12, 2010 at 8:19 PM, Mark Hulber asterisk.ad...@hulber.com wrote:
I want to use a voip speaker phone as a room monitor. Requirements:
A phone that I can set to auto answer in speaker mode.
A phone with a good speaker phone.
Ability to make the audio one way. I want to monitor
Think we need some solution WITHIN the Asterisk core. Roderick A. suggested
something that looks nice using iptables, some others have pointed out using
RBL or fail2ban, but the best would be to have some generic solution not
dependant on third party programs.
I'm not aware of the asterisk.dev
2010/4/12 Olivier oza_4...@yahoo.fr
2010/4/12 Olivier oza_4...@yahoo.fr
Hi,
In my 1.6.1.18 with dahdi 2.2.1.1, I've got :
# dahdi_hardware
pci::01:0a.0 qozap- 1397:16b8 Junghanns OctoBRI ISDN card
Does it mean I should download and use qozap or is it a bug in Dahdi ?
Have a look at:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication
It's about IAX but guess will give you some good hints on how to solve your
problem.
Alyed
2010/4/13 Mike Diehl mdi...@diehlnet.com
Hi all,
I'm trying to tighten things up a bit and I seem be be running into
On 13/04/10 00:27, Tom Stordy-Allison wrote:
Yep - this is the same codebase - the attack that I had from an EC2 yesterday
and the day before, all had the User-Agent: friendly-scanner too.
Looks like they are branching out
Go with Joshua Steins blog post - it worked perfect for me
On Mon, Apr 12, 2010 at 06:16:51PM +0200, Olivier wrote:
Hi,
In my 1.6.1.18 with dahdi 2.2.1.1, I've got :
# dahdi_hardware
pci::01:0a.0 qozap- 1397:16b8 Junghanns OctoBRI ISDN card
Does it mean I should download and use qozap or is it a bug in Dahdi ?
DAHDI 2.3.0 includes
On Tue, Apr 13, 2010 at 08:27:11AM +0200, Randy R wrote:
On Mon, Apr 12, 2010 at 7:17 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
That only addresses EC2 (and assumes that Amazon has any interest in
protecting their reputation). What about attacks that come from other
locations?
On Mon, Apr 12, 2010 at 04:58:42PM -0500, JR Richardson wrote:
Perhaps if there was a Asterisk RBL we could all contribute to; for
which we could then hook into and drop any connection where a
source IP is listed ? -- Thanks, Phil
I love the idea of a RBL... count me in for
- Original Message -
Think we need some solution WITHIN the Asterisk core. Roderick A.
suggested something that looks nice using iptables, some others have
pointed out using RBL or fail2ban, but the best would be to have some
generic solution not dependant on third party programs.
On Tue, 13 Apr 2010, Alyed wrote:
Think we need some solution WITHIN the Asterisk core. Roderick A. suggested
something that looks nice using iptables, some others have pointed out using
RBL or fail2ban, but the best would be to have some generic solution not
dependant on third party
Hello,
I have a question concerning SNOM M9 base station.
If my customer places a SNOM M9 base station in place A and a SNOM M9
base station in place B, which is 100 meters further... will a SNOM M9
handheld from base station A register to base station B when it enters
its DECT-environment.
Have you tried the SNOM forum ? They would probably have more info for
you http://forum.snom.com/
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 13 April 2010 10:12
To: asterisk-users@lists.digium.com
- Original Message -
On Tue, 13 Apr 2010, Alyed wrote:
Think we need some solution WITHIN the Asterisk core. Roderick A.
suggested something that looks nice using iptables, some others have
pointed out using
RBL or fail2ban, but the best would be to have some generic solution
Dear all,
Currently I am using asterisk 1.4.23.1. . Over the period of 1 week,
the memory in use starts off at 50% and
continues to climb until it hits 99%. When memory usage ratio become
50% or more, the quality of calls become
extremely noisy. The call quality goes back to being perfect once I
On Tue, Apr 13, 2010 at 06:42:50PM +0900, kamrun nahar bina wrote:
Dear all,
Currently I am using asterisk 1.4.23.1. . Over the period of 1 week,
the memory in use starts off at 50%
Is there much active swapping?
Run 'vmstat 1' for a while.
Look at the columns 'si' (swap in) and so (swap
Am 13.04.2010 10:47, schrieb Gordon Henderson:
I'd strongly disagree with this. (And I was the OP of this thread and had
my home/office network connection taken down due to it)
But then, I'm an old worldy Unix sysadmin and the philosophy of having a
program do one thing well is still etched
Dear Tzafrir Cohen,
Now I executed vmstat 1, Now memory usage is 15% thats why (swap in) and
so (swap out) is 0. But When memory usage become 50% or more then swap size
become 224172 kB according to previous log. May be this is the reason for
becoming sound noisy? But How i will solve this
You need to post your sip.conf and any included files in it.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-04-13 2:04 AM, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I'm trying to tighten things up a bit and I seem be be running into
something
that doesn't make
Hi all,
My Asterisk connect to GSM core network (connect directly to MSC) through E1
lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ?
Thanks in advance
--
_
-- Bandwidth and Colocation Provided
This is not actually a problem... it's a side affect of how older
versions of libpri handled PTMP links. Basically, after 3-5 minutes,
the other side is probably trying to drop layers 1 and 2 due to no calls
being active. For the most part, unless you see any issues, you should
just ignore
Hi people, I have an extension which has configured the follow me (it derives
to an IVR).
If in my dialplan I put Dial(extenX) (where extenX is that extension) and if it
is not available, it should execute the IVR, is that right?
Well, I think it should be, but it doesn't...
Here is my CLI:
On 12 Apr 2010, at 22:14, asterisk-users-requ...@lists.digium.com wrote:
There's system clock, and hardware clock.
Whatever you get for the localtime when you do 'date' command is what
you're going to get for logs from asterisk.
It seems somewhere you have your system set to run in GMT,
On Tue, 2010-04-13 at 09:47 +0100, Gordon Henderson wrote:
On Tue, 13 Apr 2010, Alyed wrote:
Think we need some solution WITHIN the Asterisk core. Roderick A. suggested
something that looks nice using iptables, some others have pointed out using
RBL or fail2ban, but the best would be to
Thanks for the input. Problem was solved by adding transfer=no in
zapata.conf
For those who need TBCT, then add transfer=yes and facilityenable=yes in
zapata.conf.
However, if your telco has RLT or TBCT as a value added service and you have
not subscribed to it then you will face my problem if
Hi,
When typing dahdi_scan on an OctoBRI-enabled setup, I've got only 8 replies
such as :
[1]
active=yes
alarms=RED
...
[8]
active=yes
...
framing=CCS
I would expect 16 replies (one per B-channel).
Is this correct ?
Regards
--
That is a function of the repeater. The repeater can manage 16 phones
and pass them back to the base station. Always look at DECT as low
power GSM because that is what it is... You have termination points
and repeaters.
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS
Hi all,
On an Asterisk/Zaptel 1.4 system, one way to gather diagnostic info is:
~# cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
Span
Hi!
Any aditional security within * is fine, but if someone is simply
drowning your bandwith, action must be taken at a lower level.
Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip,
mail, ssh, ldap, http, rsync, (or any other service you might be running)
However, I
Hi
We're using asterisk 1.4.17 using RealTime and my boss has decided that
we should keep a track of the full history of incoming calls i.e. who
and when they were transferred to. The asterisk CDR only holds the
initial answering channel for any call and not any further transfers
that may
i have installed the asterisk 1.6 before that installed the necessary
packages in Debian,
* i followed the steps as follows,
r...@astserver: ~# apt-get install unixodbc unixodbc-dev odbc-postgresql
postgresql-8.1 postgresql-contrib postgresql-dev
* then i installed the asterisk 1.6 version
On Apr 13, 2010, at 8:04 AM, Hans Witvliet wrote:
On Tue, 2010-04-13 at 09:47 +0100, Gordon Henderson wrote:
On Tue, 13 Apr 2010, Alyed wrote:
Think we need some solution WITHIN the Asterisk core. Roderick A. suggested
something that looks nice using iptables, some others have pointed out
Hi Guys,
I have been checking logs and noticed this over the last night. Should I be
concerned? and where to look for further details?
Sample:
[2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1
successfully restarted on span 1
[2010-04-13 04:31:27] VERBOSE[3844] logger.c: --
bruce bruce wrote:
[2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1
successfully restarted on span 1
It's a normal function:
*resetinterval*: sets the time in seconds between restart of unused
channels, defaults to
3600 minimum 60 seconds. Some PBXs don't like channel
Hi,
This may be no use to you if you are using 1.4 but Call Event Logging (or
CEL) that is currently in trunk should provide an easier way to do this.
All events associated with a call e.g. Answer, Hangup, Bridge start, Transfer
etc. are logged to the usual back-ends. We use postgresql via ODBC.
Speaking of all these attacks, are there any good web managed security
monitor tools for CentOS out there that can be installed on the system so
that it can give us a visual of let's multiple failed attempts against SSH
or HTTPd?
Something nice that is simple and doesn't eat a lot resources and
Hi,
Is it me or is svn.asterisk.org down ?
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Thanks, I can sleep better now.
On Tue, Apr 13, 2010 at 10:02 AM, Doug Lytle supp...@drdos.info wrote:
bruce bruce wrote:
[2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1
successfully restarted on span 1
It's a normal function:
*resetinterval*: sets the time in
Olivier wrote:
Is it me or is svn.asterisk.org http://svn.asterisk.org down ?
It is, along with issues.asterisk.org, reviewboard.asterisk.org and some
other sites. They should be back up in the next hour.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive
On Tue, 2010-04-13 at 15:49 +0200, Philipp von Klitzing wrote:
Hi!
Any aditional security within * is fine, but if someone is simply
drowning your bandwith, action must be taken at a lower level.
Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip,
mail, ssh, ldap,
On 13 Apr 2010, at 15:22, Olivier wrote:
Is it me or is svn.asterisk.org down ?
issues. too
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
On Tue, Apr 13, 2010 at 02:19:26PM +0200, Olivier wrote:
Hi,
When typing dahdi_scan on an OctoBRI-enabled setup, I've got only 8 replies
such as :
[1]
active=yes
alarms=RED
...
[8]
active=yes
...
framing=CCS
I would expect 16 replies (one per B-channel).
Is this correct ?
No.
Olivier escribió:
Hi,
Is it me or is svn.asterisk.org http://svn.asterisk.org down ?
Regards
Yep, it's down:
mig...@laptop-miguel:~$ ping svn.asterisk.org
PING svn.asterisk.org (76.164.171.230) 56(84) bytes of data.
From orc2.api-digital.com (63.238.52.42) icmp_seq=1 Destination Host
On Tue, Apr 13, 2010 at 02:53:51PM +0200, Jaap Winius wrote:
Hi all,
On an Asterisk/Zaptel 1.4 system, one way to gather diagnostic info is:
~# cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2
On Apr 13, 2010, at 7:53 AM, Jaap Winius wrote:
Hi all,
On an Asterisk/Zaptel 1.4 system, one way to gather diagnostic info is:
~# cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear
At the moment, we have a feature where if someone's sip extension is
called, we also make another call to their mobile. We use the c
option in the zap dialstring so that the user has to press # after
answering to confirm the call (this prevents things like the
answermachine grabbing the call if
- Original Message -
Speaking of all these attacks, are there any good web managed security
monitor tools for CentOS out there that can be installed on the system
so that it can give us a visual of let's multiple failed attempts
against SSH or HTTPd?
Something nice that is simple
Cool. I am just looking over splunk. Isn't that enough by it's own? or is
OSSEC needed to give it raw data? I think these two will take quite some
time to understand. Anything simpler out there as well?
Thanks,
Bruce
On Tue, Apr 13, 2010 at 10:42 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
Cool. I am just looking over splunk. Isn't that enough by it's own? or
is OSSEC needed to give it raw data? I think these two will take quite
some time to understand. Anything simpler out there as well?
Thanks,
Bruce
On Tue, Apr 13, 2010 at 10:42 AM, --[
Hi,
On Tue, Apr 13, 2010 at 03:37:37PM +0100, Julian Lyndon-Smith wrote:
At the moment, we have a feature where if someone's sip extension is
called, we also make another call to their mobile. We use the c
option in the zap dialstring so that the user has to press # after
answering to confirm
Hi!
We're using asterisk 1.4.17 using RealTime and my boss has decided that we
should keep a track of the full history of incoming calls i.e. who and
when they were transferred to. The asterisk CDR only holds the initial
answering channel for any call and not any further transfers that may
Hi!
I am trying to do a callfiel for autodialing but when I move the callfile to
outdialing folder asterisk seems like if did the call but it doesnt.
I put here my callfile and that I get when asterisk begins to do the call
If anybody has idea, pls. Tell me
TIA
Hi there,
Does asterisk keeps the master.csv open between writes? Right now I have 2
asterisk nodes sharing every configuration file (by using a distributed
filesystem) except the master.csv files. If asterisk does not keep master.csv
file open between writes, then I can share the master.csv
Hi,
A new http://misdn.org/index.php/Howto_for_Debian doc has been published
Along with http://www.linux-call-router.de/howto.html, it describes a way to
install Asterisk along mISDN V2.
Has someone experienced with it ?
Thoughts ?
Could it be a reliable path for alternate ISDN devices like AVM
Hi there,
Does asterisk keeps the master.csv open between writes? Right now I have 2
asterisk nodes sharing every configuration file (by using a distributed
filesystem) except the master.csv files. If asterisk does not keep master.csv
file open between writes, then I can share the master.csv
Hi!
to install Asterisk along mISDN V2.
Has someone experienced with it ?
Thoughts ?
Could it be a reliable path for alternate ISDN devices like AVM boards ?
If you take a look at the misdn mailing list you will see that the future
of mISDN v2 is quite uncertain - at least it was when I
On Tue, Apr 13, 2010 at 04:32:58PM +0200, Hans Witvliet wrote:
On Tue, 2010-04-13 at 15:49 +0200, Philipp von Klitzing wrote:
Hi!
Any aditional security within * is fine, but if someone is simply
drowning your bandwith, action must be taken at a lower level.
Otherwise you endup
Hi Guys,
i have a weird thing here: when using time variables (%F %T) in a shell
script, out of dial plan (particularly system() app); it displays the right
time (same as output of date), but when same variables are used in system()
application it displays a wrong time/date (ahead of 6 hours). I
Hi Guys,
i wanted to share this with u and ask for little help at the same time:
i used iptables to secure my server, so i wnet ahead and blocked avery thing
except a couple of domain protocols and UDP ports of SIP, IAX2 and that
range 15000 to 2, tested it and OK. when in production, the
Hmmm. It would seem that it would be to Amazon's advantage to jump on this
problem,
because the accounts that are performing this activity are most likely
purchased with
stolen identities, and sooner or later the charges are going to get
reversed. Either the
credit card companies are going to
You are apparently in U.S. Central Time zone.Asterisk uses the hardware
clock and system() uses the system clock, so these are probably out of sync.
Try doing
Date and
Hwclock
From a command prompt.
_
From: asterisk-users-boun...@lists.digium.com
You are apparently in U.S. Central Time zone.Asterisk uses the hardware
clock and system() uses the system clock, so these are probably out of
sync. Try doing
Date and
Hwclock
From a command prompt.
thanks, here is the output of the two clocks you mentioned they dispaly same
info (slight
I have been monitoring AMI events and realized that they don't have timestamps.
Is that standard behaviour, or is there some way to get them to
include timestamps?
I am on 1.4. Is it available on 1.6?
--
_
-- Bandwidth and
On Tue, 13 Apr 2010, khalid touati wrote:
Hi Guys,
i wanted to share this with u and ask for little help at the same time:
i used iptables to secure my server, so i wnet ahead and blocked avery thing
except a couple of domain protocols and UDP ports of SIP, IAX2 and that
range 15000 to
On Tuesday 13 April 2010 13:27:30 Danny Nicholas wrote:
You are apparently in U.S. Central Time zone.Asterisk uses the hardware
clock
What makes you think Asterisk uses the hardware clock?
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig
Hello Zeeshan/Asterisk-users
We are having a little problem in our Asterisk pbx using our A102DE, just
like Zeeshan told us about problems with zap, even if a zap channel is in
use the Hookstat is always onhook, never changes to offhook
If the line is in use or not, the behavior of the
They actually do have a timestamp, in a manner of speaking. The uniqueid
field is a pseudo-unixtime stamp.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Church
Sent: Tuesday, April 13, 2010 1:51 PM
To:
Just what I thought - guess that's the X'th time I wuz wrong today.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Tuesday, April 13, 2010 1:58 PM
To: Asterisk Users Mailing List -
On Tue, Apr 13, 2010 at 12:56 PM, Ricardo Coelho
ricardo.tch...@gmail.com wrote:
Hi there,
Does asterisk keeps the master.csv open between writes? Right now I have 2
asterisk nodes sharing every configuration file (by using a distributed
filesystem) except the master.csv files. If asterisk
Hello all,
I wonder if somebody could provide me with some advice on how to track
what looks like a bug to me:
I've got a PHP AGI script that is called whenever I dial into the system
and also whenever I issue a specific Originate() request via AMI.
The script works fine when I dial in.
DNS!! i believe it has to do with call setup and rtp protocol cause all
devices shows as sip peers at the call time, but not 100% sure. any iptables
plz :) !
2010/4/13 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Tue, 13 Apr 2010, khalid touati wrote:
Hi Guys,
Is the Originate() call using the same context as the manual Dial-In? Could
be as simple as one Answering and the other not (or not always).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leo Burd
Sent:
On Tue, 2010-04-13 at 13:59 -0500, Danny Nicholas wrote:
They actually do have a timestamp, in a manner of speaking. The uniqueid
field is a pseudo-unixtime stamp.
While correct, it's a timestamp of when the call *started*, not when the
event happened.
--
Jared Smith
Digium, Inc.
--
At least on this forum, bad help usually leads to good help???
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Tuesday, April 13, 2010 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
i believe not only today :D, but thank u anyway for the spirit of helping
people!!
2010/4/13 Danny Nicholas da...@debsinc.com
Just what I thought - guess that's the X'th time I wuz wrong today.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
The Asterisk Development Team is pleased to announce the release of
DAHDI-Linux and DAHDI-Tools version 2.3.0.
DAHDI-Linux 2.3.0, DAHDI-Tools 2.3.0, and DAHDI-Linux-Complete are available
for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
On Tuesday 13 April 2010 14:00:36 Danny Nicholas wrote:
Just what I thought - guess that's the X'th time I wuz wrong today.
The only difference between what I think you're calling the system time
(output of date) and Asterisk is that Asterisk uses a different (internal)
library to convert the
Would making timestamp=yes in manager.conf have any effect on this behavior?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith
Sent: Tuesday, April 13, 2010 2:35 PM
To: Asterisk Users Mailing List -
My derailed train of thought came from OP's mention of Centos 5.3 - I have
to do a hwclock -s on my 5.3 box at least daily to keep a reasonable time.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
What do you mean with problems on my configuration?
This is a FXO port on zapata:
signalling=fxs_ks
group=0
channel = 1
Not a FXS...can you explain to me what were you trying to say?
Message: 4
Date: Mon, 12 Apr 2010 13:14:49 -0400
From: David Backeberg dbackeb...@gmail.com
Subject:
On Tue, Apr 13, 2010 at 8:25 PM, Steve Murphy m...@parsetree.com wrote:
Hmmm. It would seem that it would be to Amazon's advantage to jump on this
problem,
I am pushing for this, please everyone who is suffering from this
problem, submit it or write to complain to Amazon and post the message
I have a problem that when one of my SIP providers has a problem the
rest of my SIP extensions and trunks stop working until either the SIP
provider fixes the problem or Asterisk stops trying to register to that
provider. Why does this happen? A single provider having problems
should not
On Tue, 13 Apr 2010, Ricardo Coelho wrote:
Does asterisk keeps the master.csv open between writes? Right now I have
2 asterisk nodes sharing every configuration file (by using a
distributed filesystem) except the master.csv files. If asterisk does
not keep master.csv file open between
On Tue, 13 Apr 2010, Leo Burd wrote:
I wonder if somebody could provide me with some advice on how to track
what looks like a bug to me:
I've got a PHP AGI script that is called whenever I dial into the system
and also whenever I issue a specific Originate() request via AMI.
The script
On Tue, Apr 13, 2010 at 11:17 AM, khalid touati khalidtou...@gmail.com wrote:
Hi Guys,
i wanted to share this with u and ask for little help at the same time:
i used iptables to secure my server, so i wnet ahead and blocked avery thing
except a couple of domain protocols and UDP ports of SIP,
check the IRQ and make sure the TDM410P has it owns IRQ.
From: Danny Dias ing.diasda...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Fri, April 9, 2010 4:52:05 PM
Subject: [asterisk-users] Problems with Fax over TDM410P
Hello my friends...
We are
On Apr 13, 2010, at 4:22 PM, Randy R wrote:
On Tue, Apr 13, 2010 at 8:25 PM, Steve Murphy m...@parsetree.com wrote:
Hmmm. It would seem that it would be to Amazon's advantage to jump on this
problem,
I am pushing for this, please everyone who is suffering from this
problem, submit it or
- I've just learned that my system now seems to work perfectly fine if I
call AMI Originate
with $channel = 'Local/%num...@vd-dial_out';
instead of $channel = 'Local/%num...@vd-dial_out/n'; // Note the
extra /n at the end
I thought it was important to use '/n' to avoid weird behavior
Hello all,
What are the possible values returned by AMI Originate when it's called
with Async set to 0?
Is there any way to find out whether the dialed channel was busy,
invalid, etc. without requiring Async to be 1?
Thanks in advance,
Leo
--
Sorry, the last message was incomplete.
So with AMI encoding the Rhino card wouldn't work reliably, on which they
were able to send us new zaptel drivers patched for our use. That fixed the
issue on our end.
Zeeshan A Zakaria
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Sent from my Android phone with K-9 Mail.
On 2010-04-13 5:25 PM,
Hi Danny,
Actually the issue I faced was the opposite, i.e. the channels would stay
offhook even after the hangup. Now I can't remember all the details but that
setup had a lot of problems, primarily because it was a very customized
system, and the Rhino T1 card was not able to correctly work
On Tue, 13 Apr 2010, Asterisk Development Team wrote:
* Static /dev/dahdi files are not generated at install time since udev is used
on all the supported distributions. build_tools/make_static_devs is
available for those users who still need the static device files.
Please do not ever
I have an Asterisk box, 1.4.30 with a PRI.
A Mitel 3300 is connected to the Asterisk box via SIP trunking.
When a user calls from the Mitel through the Asterisk box the user can speak
but can not hear the far end.
But - when I route the Mitel user to echo() it works, send and receive. The
Why do versions 1.6.2 and 1.6.1 use much more CPU resources that 1.6.0?
I can get 400+ SIP/G.711
calls running on this dual core box with 1.6.0 but the cpu maxes out and
core dumps at approx. 180 calls when version 1.6.1/2 is running.
John
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How about a generic beep detector? One that detects beeps at various
frequencies not fixed frequencies that would listen to the RTP audio stream and
send out a manager event when a detection occurs?
John
-Original Message-
Hi Jerry,
On Thu, Apr 8, 2010 at 6:54 PM, Jerry Geis
On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias ing.diasda...@gmail.com wrote:
What do you mean with problems on my configuration?
This is a FXO port on zapata:
signalling=fxs_ks
group=0
channel = 1
Not a FXS...can you explain to me what were you trying to say?
On Tue, Apr 13, 2010 at 6:55 PM, David Backeberg dbackeb...@gmail.com wrote:
On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias ing.diasda...@gmail.com wrote:
What do you mean with problems on my configuration?
This is a FXO port on zapata:
signalling=fxs_ks
group=0
channel = 1
Not a FXS...can
On Tue, Apr 13, 2010 at 06:59:01PM -0400, David Backeberg wrote:
On Tue, Apr 13, 2010 at 6:55 PM, David Backeberg dbackeb...@gmail.com wrote:
On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias ing.diasda...@gmail.com wrote:
What do you mean with problems on my configuration?
?This is a FXO port on
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