On 26.11.2010 17:29, David Backeberg wrote:
2010/11/25 Захаров Антонins...@mail.ru:
Hello everyone.
I have a timing slips errors and I can't understand what source of the
problem is.
My installation has 2 digium cards: TE420 and TE220 cards in one server.
There are 3 spans (E1) to PSTN and 3
Am 29.11.2010 08:20, schrieb Tilghman Lesher:
On Saturday 27 November 2010 04:52:31 Klaus Schwarzkopf wrote:
Hi,
why have many files on
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ the
change date 18 aug 2009? See:
asterisk-1.2.24-patch.gz07-Aug-2007 17:10
Здравствуйте.
Спасибо за ответ. Меня какраз интересуют проблемы, которые решает этот
кабель. Достать его теоретически мы сможем. Другой вопрос, в чем же его
уникальность? И почему, например, нельзя использовать floppy кабель?
Когда я поставил floppy кабель вместо официального и модулю указал
Hello
Some SOHO prospects only have a cellphone and I was wondering if
someone had investigate running Asterisk on a smartphone, to perform
tasks such as IVR, CID rewriting, voice-mail, notifications through
e-mails, etc.?
Thank you.
--
This sounds a bit suspect. What sound files are you talking about?
Voicemail? Prompts? Responses? Dictation?
Phone call recordings, outgoing and incoming to and from the call
center.
That explanation sounds bogus. Where are you seeing segmentation errors?
What processes are faulting? Do
hello,
i'm testing sending ISDN cause codes to customer pbx (test scenario for
unallocated number)
topology:
PSTN-E1-AsteriskA-AsteriskB-SOMEPBX
INVITE from SOMEPBX to PSTN
AsteriskA sends to AsteriskB
Status-Line: SIP/2.0 503 Service Unavailable
X-Asterisk-HangupCause: Unallocated
Do you mean, using the smart phone as an Asterisk server, or as a device (i.e.,
an extension)?
I think running Asterisk in server mode would run up against blocking of SIP
traffic on most voice networks. Also, you would probably run into issues with
battery life, and with availability (what if
On 11/18/2010 08:01 PM, Sebastian wrote:
Is anybody here familiar with the meaning of INVAL packets for IAX2?
Every few days I get a dropped outgoing call in the middle of the
conversation (the outgoing call has been connected for few minutes) when
an incoming call comes in. The log reads
Thank you, i want to follow your idea, how i can send and receive data from/to
Command Line in PHP Script?Thank you in advance
Date: Sat, 27 Nov 2010 08:45:47 -0800
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to hangup all channels
2010/11/27 Fabiano Carlos Heringer b...@grupoheringer.com.br
Hi, it´s possible to mantain the original CallerId when making transfers?
(atx or blind)
Example: A calls to B, A transfer to C, C see the CallerID of B, and not
A...
It´s possible?
yes
Thanks1
--
2 ways:
Read http://www.voip-info.org/wiki/view/Asterisk+AGI
or in PHP - system (asterisk -rx 'core restart now' /dev/null);
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe D'alessio
Sent: 29
Un-top-posting...
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
On Sat, 27 Nov 2010, Steve Edwards wrote:
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x
Un-top-posting...
From: Giuseppe D'alessio
Thank you, i want to follow your idea, how i can send and receive data
from/to Command Line in PHP Script?
On Mon, 29 Nov 2010, Andrew Thomas wrote:
Read http://www.voip-info.org/wiki/view/Asterisk+AGI
An AGI is executed in the context of a
On Sun, 28 Nov 2010, Jeremy Kister wrote:
On 11/28/2010 12:03 PM, Silver Thorne wrote:
So, I am wondering if anyone has a firewall/IP tables statement that
keep out unauthorised users? No one seems to get in as we use really
http://jeremy.kister.net/code/iptables/
if you already have an
Re-top-posting...
I was merely pointing out that AGI exists (teach a man to fish...)!
Sorry for not being as perfect as you...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: 29 November
On 11/29/2010 11:03 AM, Jeff LaCoursiere wrote:
If I am digesting it correctly, this set of iptables rules does exactly
what fail2ban would do, minus the logging, and without the overhead of a
scripting language, correct?
Very similar to fail2ban, but not quite the same:
* this'll block
Un-un-top posting...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: 29 November 2010 15:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
On Mon, 29 Nov 2010 02:26:35 -0800, Kevin Keane wrote:
Do you mean, using the smart phone as an Asterisk server, or as a device
(i.e., an extension)?
I think running Asterisk in server mode would run up against blocking of SIP
traffic on most voice networks. Also, you would probably run into
On Sun, Nov 28, 2010 at 12:24 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Sun, 28 Nov 2010, Silver Thorne wrote:
I have noticed lately that there have been several attempts to hack our
Asterisk server.
So, I am wondering if anyone has a firewall/IP tables statement that
keep out
So when someone's brute forcing your server is there a way to identify
the originating IPs without using a tcpdump? When I get a failed auth
on the console it shows 'acco...@asteriskserver' then tag=as25ca5023 (or
some random string, though it's a bit odd as alwaysauthreject = yes is
on in
The desired result is that user A's phone rings; when he picks it up,
user B is dialled, and user A's channel is connected to that. (This is
to be a back-end for a web-based address book.)
--
_
-- Bandwidth and Colocation
On Mon, Nov 29, 2010 at 2:01 PM, Hose hose+aster...@bluemaggottowel.com wrote:
So when someone's brute forcing your server is there a way to identify
the originating IPs without using a tcpdump? When I get a failed auth
on the console it shows 'acco...@asteriskserver' then tag=as25ca5023 (or
I have recently built a single-T1 Asterisk box using an HP DL120G6
with a Digium TE122 card.
I was finding that I was getting missed interrupts on the TE122,
causing the driver to report that it was increasing latency. It kept
doing this until the T1 did not work reliably.
I tried my usual
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub dotnet...@gmail.com wrote:
Sorry,
what I meant was:
server*CLI remove extension (hit tab)
segfault..
1.4.22
It could be an extension name Where is the error trapping if this is the
case.. Who writes this shit?
If you remove an extension that
On 11/29/2010 11:11 AM, Tony Mountifield wrote:
I have recently built a single-T1 Asterisk box using an HP DL120G6
with a Digium TE122 card.
I was finding that I was getting missed interrupts on the TE122,
causing the driver to report that it was increasing latency. It kept
doing this until
On Mon, Nov 29, 2010 at 11:08:36AM +0100, Gilles wrote:
Hello
Some SOHO prospects only have a cellphone and I was wondering if
someone had investigate running Asterisk on a smartphone, to perform
tasks such as IVR, CID rewriting, voice-mail, notifications through
e-mails, etc.?
I believe
On 11/26/2010 05:05 AM, Olivier wrote:
Hi,
On a Lenny system, with dahdi 2.4.0, libpri 1.4.11.5 and asterisk
1.6.1.18, I inserted a new Digium HA8 + B400M card.
My usual installation fails.
I can see it listed :
# lspci -n | grep d161
01:0b.0 0200: d161:8007 (rev 11)
# lspci -vn
On 11/27/2010 11:03 AM, James Lamanna wrote:
Hi,
After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC
errors on my console:
[Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad
FCS (8) on Primary D-channel of span 1
[Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c:
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub dotnet...@gmail.com wrote:
Sorry,
what I meant was:
server*CLI remove extension (hit tab)
segfault..
1.4.22
It could be an extension name Where is the error trapping if this is the
case.. Who writes this shit?
If you get hurt do you blame your
On Mon, Nov 29, 2010 at 11:07 AM, Roger Burton West ro...@firedrake.orgwrote:
The desired result is that user A's phone rings; when he picks it up,
user B is dialled, and user A's channel is connected to that. (This is
to be a back-end for a web-based address book.)
This is click-to-call.
On Mon, 29 Nov 2010, Gilles wrote:
Hello
Some SOHO prospects only have a cellphone and I was wondering if
someone had investigate running Asterisk on a smartphone, to perform
tasks such as IVR, CID rewriting, voice-mail, notifications through
e-mails, etc.?
While I can run Asterisk on my
On 29 November 2010 18:52, C F shma...@gmail.com wrote:
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub dotnet...@gmail.com wrote:
Sorry,
what I meant was:
server*CLI remove extension (hit tab)
segfault..
1.4.22
It could be an extension name Where is the error trapping if this is
the
On 11/25/10 7:12 AM, Jonas Kellens wrote:
@ Shaun Ruffell : What do you mean by Wall time ?
This server is indeed also time server (ntpd is running)
Basically that the time on the server matches up with the time actual
time you would see on a wall clock. Based on your response to Willaim
r...@pbx:~# uptime
23:10:15 up 606 days, 9:38, 1 user, load average: 0.31, 0.08, 0.02
Customer called they are having a scheduled power outage for most of
the day because of construction if I can shut down the machine
gracefully. So I decided to run uptime first.
Enjoy
--
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