Hi All,
I'm not the first to try to start a VOIP blacklist but currently working on
a project for the next 12 hours, hopefully I can get it up soon. What I
intend to do is to work with a few reliable Harvester to gather the logs. A
simple script to parse it then extract the list of attackers IP,
What determines how long SIP channel waits, when you dial a peer with no
registration, before returning ${DIALSTATUS} CONGESTION?
When I dial a peer with no registration, SIP channel currently waits
many seconds before returning ${DIALSTATUS} CONGESTION - how can I
shorten this timeout?
--
On Tue, May 04, 2010 at 08:46:39AM +0100, Steve Howes wrote:
On 4 May 2010, at 03:44, Jack Bates wrote:
We recently got VoIP, so when we make a call, Asterisk should first try
to make the call with VoIP, but in case either our VoIP or our internet
service are down, Asterisk should then try
How do you configure Asterisk to dial, in order, each channel from a
group of channels until it either finds an available channel, or runs
out of channels?
We recently got VoIP, so when we make a call, Asterisk should first try
to make the call with VoIP, but in case either our VoIP or our
I've got several SIP clients with dynamic IP addresses
Asterisk has one public and one private IP address
SIP clients might connect to Asterisk from either the internet or the
private network (192.168.1.255) - they're portable
By default, directmedia/canreinvite is enabled and Asterisk sets up
I configured our SIP gateway to automatically dial extension s when a
phone is picked up. I want Asterisk to play a dial tone, wait for an
extension to be dialled, and hangup on timeout
This works great, but I also want Asterisk to *stop* playing the dial
tone after the first digit is pressed
So
I want the first line of my dialplan to check and expression, and exit
from the dailplan if it is true - is there a convention for this?
My goal is to exit from the dialplan before calling Answer() if the
callerid is null. By this means I hope to work around this issue:
I am using Asterisk and an X101P card as a glorified answering machine.
We have a residential PSTN line with about six phones connected to it.
Like an answering machine, I want Asterisk answer the line *only* when
an incoming call is not answered after four rings.
This mostly works. My
active channels
1 active call
Asterisk version is 1.4.10.1 with FreePBX 2.3.0.3.
Thanks for your help.
Regards, Jack
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Tzafrir Cohen schrieb:
On Mon, Jul 30, 2007 at 02:01:49PM +0200, Jack wrote:
Hi all,
I have a Wildcard TE110P connected to a E1 line an I want to reserve
channels in the following way:
channels 1-15 and 17-21 for incoming calls
channels 22-28 for outgoing calls
channels 29-31
Eric ManxPower Wieling schrieb:
Jack wrote:
Hi all,
I have a Wildcard TE110P connected to a E1 line an I want to reserve
channels in the following way:
channels 1-15 and 17-21 for incoming calls
channels 22-28 for outgoing calls
channels 29-31 for emergency calls
My zaptel.conf
).
You mention good points and although I (in the moment) have no need to
think about wasted channels and wasted money I will keep this in mind.
On 7/30/07, Jack [EMAIL PROTECTED] wrote:
Hi all,
I have a Wildcard TE110P connected to a E1 line an I want to reserve
channels in the following way
Tzafrir Cohen schrieb:
On Tue, Jul 31, 2007 at 10:06:30AM +0200, Jack wrote:
Tzafrir Cohen schrieb:
On Mon, Jul 30, 2007 at 02:01:49PM +0200, Jack wrote:
signalling=pri_cpe
channel = 29-31
and then in extensions.conf:
[hangup-calls]
; not sure
context=from-zaptel
channel = 1-15
channel = 17-21
; outgoing
group = 2
signalling=pri_cpe
channel = 22-28
; emergency
group = 3
signalling=pri_cpe
channel = 29-31
How can I avoid that incoming calls are going to the channels 22-31?
Regards, Jack
Carsten Bock schrieb:
José Luis Ledesma schrieb:
In my asterisk 1.4.5 chan_features.so has been installed properly...
check in your asterisk-source if /channels/chan_features.so is present
regards,
Jack escribió:
Is chan_features.so deprecated for asterisk 1.4.5 or why
Joshua Colp schrieb:
Jack wrote:
Thanks for your answer and sorry for my late response.
So what does this exactly mean to me? Can I keep chan_features.so from
1.4.4? What consequences does it have when chan_features.so is disabled
und why has this been done? Is chan_features.so related
before
attempting to run Asterisk.
Is chan_features.so deprecated for asterisk 1.4.5 or why is this
module not installed by asterisk 1.4.5?
Regards, Jack
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Hi,
has anyone managed to get hudlite server working on a Debian Etch based
installation of Asterisk 1.4?
So far I managed to eliminate all error messages, but the process is
killed directly after starting the hudlite server without showing any
error messages.
I would be very happy if
Hi,
I have the problem that WaitExten is not responding to key presses. Here
are the sections from my extensions.conf:
[globals]
incoming_call=0
menu=0
announce=0
[internal]
exten = 777,1,Goto(hotline,${EXTEN},1)
[hotline]
exten = _X.,1,Set(CALLERID(name)=Hotline)
exten =
Hi,
has anyone managed to get hudlite server working on a Debian Etch
based installation of Asterisk 1.4?
So far I managed to eliminate all error messages, but the process is
killed directly after starting the hudlite server without showing any
error messages.
I would be very happy if anyone
on
'SIP/5060-08c53e68'
-- Executing [EMAIL PROTECTED]:1] Hangup(SIP/5060-08c53e68, ) in new
stack
== Spawn extension (play_recording, h, 1) exited non-zero on
'SIP/5060-08c53e68'
Does anyone tell me why this is happening?
Thanks,
Jack
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asterisk-users@lists.digium.com 12/15/06 11:25
On Friday 15 December 2006 10:52 am, Eric ManxPower Wieling wrote:
DACS is not done in Asterisk. DACS is done in the Zaptel drivers. In
fact, you can do DACS without Asterisk even being installed on the
Asterisk from SVN-trunk-r44731. Any
help?
Thanks,
Jack Morgan
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On Tuesday 17 October 2006 11:12, Jack Morgan wrote:
All,
I'm not able to play background files since this morning. I'm seeing this
error message in the logs:
[Oct 17 10:23:56] WARNING[4572] file.c: File
custom/asterisk-prospectus_IVR-main-day does not exist in any format
[Oct 17 10:23:56
be appreciated!
Thanks,
Jack Morgan
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Hi,
Is there way a way to restrict access to certain menus, such as the
following:
0 Mailbox options
1 Record your unavailable message
2 Record your busy message
3 Record your name
4 Record your temporary message (new in Asterisk v1.2)
Thanks in advance,
Jack
. The
max calls I can achieve is 200 simultaneous but audio is really chopping
due to high jitter. Does anyone know how to optimize Asterisk and/or
RedHat Enterprise Linux 4 to increase simultaneous calls?
Thanks,
Jack
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http://www.voip-info.org/
look for Asterisk link on the left.
Danko Miocevic wrote:
Where can I get some asterisk books.. or tutorials..? I´ve been
searching in google.. but I find just some tutorials explaining how to
fast set up an asterisk server. I want to learn how to use it and how
to
.
Jack
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Hi all,
I have 2 servers and I'm trying to configure iax to call from
Server2 (fxo) to Server1 (sip extension)
Server1: 2 sip's extension (123 and 321)
Server2: TDM-400 1 fxo(extension 999, channel 3) and 1 fxs (channel 4)
from 123 to 321 it's all right in both ways
from 999 to 123 no audio
This seems to be due to a driver conflict. If I unload Zaptel, the
sound returns.
I'm having the same issue with a 2.4 kernel on whitebox 3 using HEAD.
Still investigating... let me know if you find anything new.
Jack
On 6/29/05, Jeremy McDermond [EMAIL PROTECTED] wrote:
I've looked all
I'm having the same issue. If I unload Zaptel, and restart
asterisk... the sound does return.
On 7/25/05, Arnd Vehling [EMAIL PROTECTED] wrote:
Hi,
i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07
and everything worked fine sofar when suddenly the voicemail and
are playing and I'm getting no errors so its a bit of
a mystery.
-Jack
On 7/4/05, RockWater ! [EMAIL PROTECTED] wrote:
Hello anyone who can help
I have two Asterisk boxes with identical hardware (Dev Production). I
recently rebuilt the DEV box using Fedora Core 3 and the latest CVS Head
I installed Asterisk 1.0.9 in a Freebsd 5.4 ( with no zaptel card); I
have 2 zoom x5v and works great ( in extensions 123 and 321 ) but I
was trying to test cmd Playback, MusicOnHold, MP3Player but when I
call to extension 100 I don't hear the sound ( mp3 or gsm that I put)
, I only hear noise
If
I have installed asterisk in a 4.11 RELEASE FreeBSD, and we are using
two Zoom X5v with SIP and works fine, we can call each other and this
is OK
--extensions.conf--
[general]
static=yes
writeprotect=no
[sip]
exten = 123,1,Dial(SIP/123,20)
exten = 123,2,Voicemail(u${EXTEN})
=asterisk
attach=yes
maxmessage=180
pbxskip=yes
fromstring=The Asterisk PBX
[default]
123 = 123,peter,[EMAIL PROTECTED],attach=yes
321 = 321,jack,[EMAIL PROTECTED],attach=yes
2. When I try to play an mp3 with MP3player in extensions.conf I only
hear noises
Any help
Hi Kerry,
I've been reading your writeups and awaiting each new installment. This
project is new to me from your articles, and as my wife will tell you, I
have been spending lots of after-hours time working on setting this up
since then. :) I suspect you and the link on slashdot have bumped
... use a different number for
each and if you don't want the area code change the digit pattern to
reflect it... so that after the 8, it fits the pattern or just use _8.
to accept the whole string.
hope this helps,
Jack
On Fri, 15 Oct 2004 21:04:55 +0200 (CEST), Remco Barende
[EMAIL PROTECTED] wrote
, but it doesn't seem appropriate
for what I want to do above.
Any ideas?
Thank you!
Jack
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Any thoughts on the following?
I am running asterisk from CVS (downloaded yesterday's
version, just to be sure) on a test system with no
digium cards in it, so I have installed ztdummy (see
logs and screenshots below) as a timing source.
When I call the music on hold extension from a Sipura
Sip
Does any combination of Asterisk hardware and software
exist that supports connection to the Japanese
telephone system? (T1, E1 or J1 would be preferable,
but analog would be OK as well as a last resort (20
lines though..)).
Anyone have any thoughts or experience with this, (or
if Asterisk is
(copy_via_headers):
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Thanks
Steven
*
Steven Jack
University of Glasgow
Computing Service
Glasgow G12 8QQ
[EMAIL PROTECTED]
Tel +44(0)141 330 3828 Fax +44(0)141 330 3820
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