To be honest this is the first time I see this wiki mentioned. It doesn't
even come up in talks on this list. The wiki should be advertised often and
there should be some sort of active monitoring and supervision of the
contents as well as some serious ongoing official contributions. All this
well
That fixed it! THANK YOU.
-Cassius
From: Peder pe...@networkoblivion.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wed, 24 Nov 2010 07:42:52 -0600
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Premature reply. It did fix the first issue. Now when I ring that phone I
get busy here from the phone, and the call goes straight to voicemail per
dialplan. Maybe another parameter in addition to Reorder Delay?
From: Cassius Smith cass...@cassius.org
Date: Thu, 25 Nov 2010 10:34:25 +0100
To:
Hi all,
I want to use asteriks with my siemens hipath 1120 hardware.
Is it possible?
Thanks
--
/**
* @AUTHOR Atıf CEYLAN
* Software
Developer System Admin
* http://www.atifceylan.com
*/
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_
-- Bandwidth and
The proble is dialplan I configure fine
--
Sent from my BlackBerry®
VoIP, Windows/Linux Administration and Network Management
US Numbers: 561-886-0664
Nicaragua Mobile: +505.8488.6876
-Original Message-
From: Stefan Schmidt s...@sil.at
Sender: asterisk-users-boun...@lists.digium.com
Hello.
Thank you for the feedback.
To reply to all the information :
@ Shaun Ruffell : What do you mean by Wall time ?
This server is indeed also time server (ntpd is running)
@ Mark Deneen : No, no monitor attached. This is a Xen VPS. I do have a
VPS interface, but this is also frozen when
On Tue, Nov 23, 2010 at 06:51:37PM -0500, John Novack wrote:
You should also have, in general:
alwaysauthreject=yes
This seems pretty effective in stopping some hacking
These are simple fixes.
I found it very effective to make sure the handled sip domains don't
contain the ipadress(es) of
On 25 November 2010 13:02, bayardo.sanc...@gmail.com wrote:
The proble is dialplan I configure fine
--
Sent from my BlackBerry®
VoIP, Windows/Linux Administration and Network Management
US Numbers: 561-886-0664
Nicaragua Mobile: +505.8488.6876
-Original Message-
From: Stefan
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 internal PBS stations -
normal installation for transit communication.
Span
Stefan,
I had the same problem. And I upgrade to 1.4.37 to resolv.
Leonardo Silva
2010/11/25 bayardo.sanc...@gmail.com
The proble is dialplan I configure fine
--
Sent from my BlackBerry®
VoIP, Windows/Linux Administration and Network Management
US Numbers: 561-886-0664
Nicaragua Mobile:
Hi,
I'm testing an upgrade from 1.4.18 to 1.4.37 in a VM prior to putting it
into production.
Ive done this by installing 1.4.18 onto the VM, putting my config files
in place and then installing 1.4.37 over the top (which is what I'd have
to do on production).
I've found a few issues in the
I am studying about echo cancellation in asterisk and I want to use the
numeric information from dahdi_monitor verbose for my research.
Unfortunately, I couldn't find anything about the unit of measurement used
in this tool. Which unit is used to measure the signal level?
--
Hi Gary,
I went through this process a few times over the past few years.
Theres a few short guides for securing Asterisk, but much of it depends
on your design. If it's a traditional POTs-type PBX then locking down
IPs using firewalls is a great thing, however if you make use of
inbound-SIP
One thing we did to secure remote users is to use SNOM370s and OpenVPN..
--
Singer XJ Wang, Senior System and Database Administrator
The Pythian Group - love your data
http://www.pythian.com
Desk: (613) 565-8696 x298
Cell: (613) 266-3763
On Thu, Nov 25, 2010 at 12:33, Adrian Marsh
Michael Smith wrote:
Starting in Asterisk 1.8.0, Asterisk supports connected line updates.
This is fantastic for SIP. How can I prevent them from being sent to a
PRI channel?
Looking through the chan_dahdi and sig_pri code, I don't see any
configuration flag to block the updates from
Hi all,
Anyboby has some experience with D-link dvg-3032s with asterisk and could
give me some support.
I am using this dial plan :
exten = _X. ,1,Dial(SIP/${EXTEN:0...@192.168.0.60,30)But I receive the mesagem
from C.O that the number is incorrect.
And about incoming calls, I put I
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