On Wednesday 02 January 2013, Frank wrote:
Greetings all,
I have been seeing a lot of
[Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite:
Sending fake auth rejection for device
100sip:100@108.161.145.18;tag=2e921697
in my logs lately. Is there a way to automatically
I am using fail2ban on all my asterisk server, but beware, fail2ban can be
a dangerous software. The problem rely on the fact that SIP uses UDP, so it
is possible to send messages with a forged source IP address. This way the
bad guy out there can ban all your IP addresses. I say it is possible
2013/1/3 bilal ghayyad bilmar...@yahoo.com
Hi;
How can I know the duration that the DAHDI channel is still used? I need
to know its status and since when it is in this status, how?
Also, is it possible to hangup the channel if it has been openned more
than 90 minute? Other than using the
On Thu, 2013-01-03 at 09:42 +0100, Leandro Dardini wrote:
I am using fail2ban on all my asterisk server, but beware, fail2ban
can be a dangerous software. The problem rely on the fact that SIP
uses UDP, so it is possible to send messages with a forged source IP
address. This way the bad guy
On Thursday 03 January 2013, Selva M wrote:
Hi,
I setup PBX with A400P 4 x FXo board. There are one analog line plugged
into port 1.
Internal extension cane make calls to PSTN without any issue.
When I make inbound call, caller get busy tone user busy' message right
away.
Wow! Thanks so much for all the information. I now have a lot to look over.
Bob R
On 01/02/2013 10:03 AM, Tzafrir Cohen wrote:
On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote:
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
I don't think this should be an issue, but we have seen a lot of sites
going live and discovering too late that they had recording problems. Maybe
you won't need to implement an external recorder, but it's better to plan
in advance, not when you are in production! :)
l.
2013/1/2 Leandro Dardini
On Wednesday, January 2, 2013, Frank wrote:
Is there a way to automatically ban IP address from
attackers within asterisk ?
As others have mentioned, fail2ban does a good job. However, it may
not be enough as these attacks sometimes come from older versions of
the SipVicious hacking tool that
Interesting...
-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de Geoff Lane
Enviada em: quinta-feira, 3 de janeiro de 2013 10:06
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re:
All,
We are in the process of trying to setup our network to use Verizon's SIP
trunking product. They say that since Asterisk is not on their certified
list of approved devices, we need to go through a field trial to get it
approved before allowing us to use their service.
Where we are at is
On 3 Jan 2013, at 15:13, Michael L. Young wrote:
So, I am asking the community for any input. I have read on here and seen on
IRC that some in the community are successfully using Asterisk with Verizon
SIP. Verizon was going to check and see if they have any notes about that
and those
- Original Message -
From: Steven Howes steve-li...@geekinter.net
I *think* Verizon require IPSEC for the signalling, so it may be
worth reading up on configuring IPSEC in Linux (or acquiring
additional hardware) whilst you're looking at the Asterisk part.
This could have just been
2013/1/3 Steven Howes steve-li...@geekinter.net
On 3 Jan 2013, at 15:13, Michael L. Young wrote:
So, I am asking the community for any input. I have read on here and
seen on IRC that some in the community are successfully using Asterisk with
Verizon SIP. Verizon was going to check and see
It doesn't matter. They still require IPSEC VPN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael L. Young
Sent: Thursday, January 03, 2013 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial
Is it difficult to publish a build asterisk.deb compiled for VIA
C3 architecture ? Instead of using the binary just for me.
So any one trying to install it on C3 CPU will need just to do:
aptitude install asterisk
The one that is installed by default doesn't work for such a CPU
Should I
On Thu, Jan 3, 2013 at 8:13 AM, Michael L. Young myo...@acsacc.com wrote:
Where I am at is that they want us to use an SBC. One engineer asked
about Cisco Call Manager. I told them that basically if I can accomplish
the same thing with a Linux box (routing box and sip proxy box) without
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Thursday, January 03, 2013 2:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI: How to know since when it is used?
On 01/03/2013 09:56 AM, Carlos Alvarez wrote:
On Thu, Jan 3, 2013 at 8:13 AM, Michael L. Young myo...@acsacc.com
mailto:myo...@acsacc.com wrote:
Where I am at is that they want us to use an SBC. One engineer
asked about Cisco Call Manager. I told them that basically if I
can
The Asterisk Development Team has announced a security release for Asterisk 11,
Asterisk 11.1.2. This release addresses the security vulnerabilities reported in
AST-2012-014 and AST-2012-015, and replaces the previous version of Asterisk 11
released for these security vulnerabilities. The prior
Hello Everyone,
Before getting into SIP and RTP traces, I wanted to clarify some of
the sip.conf settings that may to some seem redundant or have a
misconception with. I do apologize if this has been discussed time and
time again as I would imagine. If anything, this email would make
google
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, January 03, 2013 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Moving User Agent To
On Wed, Jan 2, 2013 at 5:39 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi;
How can I know the duration that the DAHDI channel is still used? I need
to know its status and since when it is in this status, how?
Also, is it possible to hangup the channel if it has been openned more
than 90
Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008
Just for testing purposes, and deduce my way from there? Right now I
am trying to call the phone from my softphone. That being said, I
currently I am not able to reach the remote
Am 03.01.2013 21:21, schrieb Nick Khamis:
Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008
do you mean 1_000_8 ?
Markus
--
_
-- Bandwidth and Colocation Provided
On 01/03/2013 02:23 PM, Markus Weiler wrote:
Am 03.01.2013 21:21, schrieb Nick Khamis:
Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008
do you mean 1_000_8 ?
Markus
I think he means 10007.
--
On Thu, Jan 3, 2013 at 2:21 PM, Nick Khamis sym...@gmail.com wrote:
[Dec 12 15:35:54] WARNING[1736]: app_dial.c:2198 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
Can you check that the registration is happening correctly? Try `sip show
peers` or `sip show peer
I have been seeing a lot of
[Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite:
Sending fake auth rejection for device
100sip:100@108.161.145.18;tag=2e921697
in my logs lately. Is there a way to automatically ban IP address from
attackers within asterisk ?
You may want
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker
Sent: Thursday, January 03, 2013 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Moving User Agent
Oooops yes of course 10004-10007!! Simple math does not come easy
anymore... Anyhow, I singled out Opensips and I have two way audio
form UA(local) - UA(remote) but not from UA - Siptrunk. That being
said maybe a small diagram of the architecture. Please don't laugh!!!
:) I know having a block of
Just for grins, run netstat -anp on the call using just Asterisk and then
again with OpenSIPS in the mix. It sounds like OpenSIPS or your RTPproxy is
block the audio channels.
--
_
-- Bandwidth and Colocation Provided by
To Answer Some of You Questions:
Please not that I replace the true domain wtih example, and the true
ip for the remote UA with public-ip. Nothing against no one here,
just don't know who else would read this email in the future!!!
PS: The public IP of the remote UA is correct.
SIP Show Peers:
Hello,
We want the ability to choose from an AGI script whether or not to enable
faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone
suggest a workaround?
Thanks for any advice.
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Thursday, January 03, 2013 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] faxdetect on/off on the fly?
Hello,
We
On Thu, 3 Jan 2013, David Cunningham wrote:
We want the ability to choose from an AGI script whether or not to
enable faxdetect for calls over SIP or DAHDI.
What's the 'use case?'
You're going to call in and execute an AGI that will enable faxdetect for
future calls to this channel or other
On 03/01/2013 11:04 AM, Jeff LaCoursiere wrote:
On 01/03/2013 09:56 AM, Carlos Alvarez wrote:
On Thu, Jan 3, 2013 at 8:13 AM, Michael L. Young myo...@acsacc.com
mailto:myo...@acsacc.com wrote:
Where I am at is that they want us to use an SBC. One engineer
asked about Cisco Call
Michael L. Young wrote:
I should have probably stated that this is going to be going through
an MPLS network being setup with Verizon. They may not be requiring
that since it is within their network, not going over the internet.
They have not said anything about the the need to secure the
Hi,
I tried the option and got following message.
PBX1*CLI
-- Starting simple switch on 'DAHDI/1-1'
== Starting DAHDI/1-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/1-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing
Hi Steve,
We have all calls going to an AGI, which decides where the number will get
routed to, and if fax detection should be enabled for this call. The choice
should only apply to the current call.
Thanks very much.
On 3 January 2013 17:46, Steve Edwards asterisk@sedwards.com wrote:
On
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