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
--
___
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
St
On 29 November 2010 18:52, C F wrote:
> On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub 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
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
On Mon, Nov 29, 2010 at 11:07 AM, Roger Burton West wrote:
> 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". It can be done
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub 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 Mama for ha
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_dahd
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)
>
> #
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 beli
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
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub 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 is being used
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 proced
On Mon, Nov 29, 2010 at 2:01 PM, Hose 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
> some random string, though it'
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 Provide
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 sip.con
On Sun, Nov 28, 2010 at 12:24 PM, Steve Edwards
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 unauthorised us
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 in
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 ho
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-u
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 20
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 h
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
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 -
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 No
2010/11/27 Fabiano Carlos Heringer
> 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
>
>
> --
> _
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 channel
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 rea
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
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 (unassign
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
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.
--
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
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-2
On 26.11.2010 17:29, David Backeberg wrote:
2010/11/25 Захаров Антон:
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 spans to interna
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