We have a small office installation running over a cable modem. (8M down, 500k
up confirmed with numerous speed test sites)
When a single call is up, call quality is fine. When a second call is up,
outbound audio is immediately choppy. We're using ulaw, and confirmed that
traffic with 2
Jitterbuffer affects inbound audio only, not outbound (the other side hears the
choppiness) so I don't think that will help/
Trunking only reduces overhead after 4+ calls, so that shouldn't help either.
(Since this occurs at 2 calls)
I can't wireshark the other end since the other end is my
Is the a CLI command that shows all channels in use at one time? (Whether IAX,
SIP, SCCP, etc)?
As well, when I SIP SHOW CHANNELS I see phones registering showing as
channels in use. Is there a way to filter this output?
Thanks!
MD
--
We have an application that plays a variety of sound files on one leg of a call
(generated by a call file). We've been told that the party listening to the
audio files intermittantly hears robotic sounding audio (on/off during the
same call).
Anyone have ideas on cause? These calls are on an
I've searched through the wiki but I can't find what I need...I'm trying to
figure out what the max call duation is. I found references to show
application AbsoluteTimeout but that isn't in 1.6 (not even prepending core
to the front). A core help show didn't help...
--
, 2011 2:44 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Dialplan to bridge 2 legs?
On Sun, 23 Jan 2011, Michelle Dupuis wrote:
Is it possible to have a call file enter the dialplan, and then initiate
2 outbound calls and then bridge them?
A call file can specify a channel
to bridge 2 legs?
Un-top-posting...
On Sun, 23 Jan 2011, Michelle Dupuis wrote:
Is it possible to have a call file enter the dialplan, and then
initiate 2 outbound calls and then bridge them?
On Sun, 23 Jan 2011, Steve Edwards wrote:
A call file can specify a channel and a context/exten/priority
Take a look at High Availability ASTerisk (HAAST) from www.generationd.com
Their software sits between the OS and asterisk, and can failover servers,
switch IP addresses, control external interfaces, etc.
It can run on different hardware (make a cluster from different/cheap boxes),
it allows
I found some great pieces of script on the internet that I've combined to allow
Asterisk to send voicemails as an MP3 file, and encode the sender name and
number as well as message number as tags into the MP3 file. I even include a
cover art image which has our company logo and PBX symbol in
Ok - I've put the script up on the www.generationd.com web site. Just go to
the Downloads | Asterisk section to pull it down.
I would like to keep control of this script so please send me changes (don't
repost elsewhere) and I'll keep the latest version up for everyone. I'll add a
link to
I have a situation where an Asterisk server is NATted, sitting behind a PIX.
One public IP is used for one purpose, now a second public IP is required for
another.
Is there a way to have Asterisk use more than one public IP when behind NAT?
(I already use the externalIP setting)...
If not,
: Re: [asterisk-users] Multiple public address to one Asterisk
serverbehind NAT?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Tuesday, February 22, 2011 3:34 PM
To: Asterisk Users
For the High Availability part check out the HAAST add-on for Asterisk at
www.generationd.com
It detects a variety of failures, shuts down the failing system, starts
asterisk on the peer, moves the IP over, etc. Runs with every Asterisk variant
and every Linux distro. No special hardware
Use simple RJ45 (8 wire) A-B switched controllable by serial port, and use
HAAST to throw the A-B switch to reroute the PRI.
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Adolphe Cher-aime
or point me
to any document of website.
--
Sent from my iPhone
On Apr 30, 2011, at 12:09 PM, Michelle Dupuis mdup...@ocg.ca wrote:
Use simple RJ45 (8 wire) A-B switched controllable by serial port,
and use HAAST to throw the A-B switch to reroute the PRI
...@lists.digium.com] On Behalf Of Kaushal Shriyan
[kaushalshri...@gmail.com]
Sent: Saturday, April 30, 2011 11:03 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] HA Asterisk
On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
There are lots out there, but here's
...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 30, 2011, at 8:31 PM, Kaushal Shriyan wrote:
On Sun, May 1, 2011 at 8:48 AM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
Yes that's it - one PRI line in, 2 out (one to the PRI card in each server).
If you have lots of PRI lines, you may
Yes - the USB connection carries the data. Keep in mind that the HA aspect
of this product just means you can connect to two asterisk servers. There is
not data replication, detection of asterisk failure, etc. (without buying more
xorcom products). Be sure to do your homework. But they do
I think the OP's point was that open source should mean:
Free to modify
Free to contribute code
Free to use.
Leaving the first two but taking away the free to use really takes the F out
of FOSS. There have been other posts discussing Digium's license requirements,
code ownership, etc. I
I'm looking for recommendations for standalond PRI to SIP converters. (Needs
to be outside the asterisk box - so a PCIe card won't do)
I've used redfone but this project doesn't need the redundancy features...
Thanks!
--
_
--
Cool topic!
Our company (generationD) developed some CID scripts for free use, and we would
be interested in building and hosting this service.
On the spec side, how do we avoid users claiming numbers belonging to others?
(Could be an admin nightmare)
Do we allow number ranges?
Do we require
We ran into this a few years ago. Polycoms and Grandstreams worked fine with
#xxx extensions, but Aastra's would not. Could not dial extensions beginning
with #
We chased Aastra tech support for 2 weeks. They acknowledge the bug, and we
were told they would fix this in their next firmware
If you check the archives you might find the original messages on this topic
from a few years ago...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
[davies...@gmail.com]
Sent: Wednesday,
We are building an app that will initiate outbound calls using .call files, and
each call can be a different duration (eg: 1min to 5min). These calls will go
through an Asterisk service with other calls/apps running.
I need to control the MAX number of channels in use so I don't overload this
Has anyone written a C wrapper to ease development with the AMI? I found a
couple of c++ ones, but not C.
Thanks!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
If one server is supposed to carry the full load of the other during failure,
then you have to size each server to handle 100% load - so load balancing is
pointless.
Checkout haast at www.generationd.comhttp://www.generationd.com and read the
docs on how it does failover...certainly good for
Although you say SIMPLE...not all virtualization hosts allow software
installation. On VMware the host has become an appliance you can't really mess
with...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On
VMware is moving all server products to their ESXi engine. (The old VMware
server and ESX products are moving to legacy status - with these you could
actually do stuff on the kernel). ESXi is no longer a kernel you can mess
with, can't install drivers, etc. ESXi is being treated as an
There is a script on www.generationd.com designed for Asterisk. It will
convert the Wav49 to mp3, add call info into MP3 tags, add a company logo, etc.
and then email the message.
It's a one line change to add to asterisk - very handy. (We use it for Android
phones, nice to see call info
] On Behalf Of jon pounder
[j...@inline.net]
Sent: Friday, November 25, 2011 8:03 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] android won't play wav49: how to change format
On 11/25/2011 06:39 PM, Michelle Dupuis wrote:
There is a script on www.generationd.com designed for Asterisk
...@lists.digium.com] On Behalf Of Andrew Furey
[andrew.fu...@gmail.com]
Sent: Wednesday, December 28, 2011 11:37 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Interesting attack tonight fail2ban them
On 29 December 2011 12:07, Michelle Dupuis mdup...@ocg.ca wrote:
I thought that it might be worth
Here is more of a SIP debug log:
As you can see Asterisk retries four times but I assume the softphone is not
responding?
---
Really destroying SIP dialog
'637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4'mailto:'637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4'
Method: OPTIONS
Reliably
The BB is using wifi, on the same subnet as the asterisk server so no need for
NAT.
There is no keep alive option on the softphone (very simplistic settings)
Thanks
--
_
-- Bandwidth and Colocation Provided by
I happened to be in the cli tonight as some (208.122.57.58) initiated a simple
attack - just trying to make long distance calls from outside context.
Although harmless, this went on for several minutes as the idiot just used up
my bandwidth with SIP messages. Here's and example:
[2011-12-28
On Wed, Dec 28, 2011 at 11:07 PM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
I happened to be in the cli tonight as some (208.122.57.58) initiated a simple
attack - just trying to make long distance calls from outside context.
Although harmless, this went on for several minutes
I have a softphone I'm trying on a blackberry, that registers on my Asterisk,
can make outgoing calls, but can't receive calls.
There is very little traffic with this phone (see debug below - as the phone
registers), and sip show peers confirms it is unreachable.
Any suggestions? Is this just
1. I checked the log and I don't see any registration attempt, so I *assume*
they simply send an invite, and so they are in the external/outside context of
my dialplan. So they are trying to reach extensions which don't exist. If
they succesfully registered they would be on the internal
Wow - nice! A few quick questions:
1. How long can the recording be for translation?
2. Any limitation on how much text the return (transcribed) variable can hold?
3. Any commercial / terms of use limitations?
From: asterisk-users-boun...@lists.digium.com
We have a multitenant Asterisk 1.4 installation for multiple small business,
and we need to report how many calls a single business has active at one time.
Is there a way to VIEW how many calls are up in a single context? (Or some
other way to accomplish the same)?
Thanks
--
I have an ATCOM ATA that is trying to connect to an asterisk server using IAX.
The ATA and Asterisk are on the same subnet, not firewall/nat etc.
Below is a a log excerpt, showing the REGREQ received, and then Asterisk goes
on to send lots of REGAUTH...and this continues for a while, but the
We have an Ast 1.6 installation which is connected to an Avaya using ooh323.
Something is causing the log to fill with In ooEndCall call state is -
OO_CALL_CLEAR (incoming, ooh323c_1) messages every 100ms. This causes the log
to grow to 300MB in just 5 minutes, which eventually overloads the
you,
Vladimir
On 6/5/2012 8:58 AM, Michelle Dupuis wrote:
We have an Ast 1.6 installation which is connected to an Avaya using ooh323.
Something is causing the log to fill with In ooEndCall call state is -
OO_CALL_CLEAR (incoming, ooh323c_1) messages every 100ms. This causes the log
We've deoplyed a number of pure VoIP wireless (wifi proprietary) phones, but
not dect.
Is there a simple overview of integrating DECT phones with Asterisk somewhere?
I assume the DECT basestation has a multi-account SIP VoIP interface, and the
handsets are just plain old dect?
Can you push
: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
[car...@televolve.com]
Sent: Friday, June 29, 2012 4:58 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Intro to DECT vs IP
On Fri, Jun 29, 2012 at 1:22 PM, Michelle Dupuis
mdup
-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
[aster...@lists.minotaur.cc]
Sent: Friday, June 29, 2012 6:27 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Intro to DECT vs IP
On 29/6/12 11:16 pm, Michelle Dupuis wrote:
Can you
...@lists.minotaur.cc]
Sent: Friday, June 29, 2012 8:22 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Intro to DECT vs IP
On 30/6/12 12:12 am, Michelle Dupuis wrote:
I like the look of the C610H. Is there a matching DECT base station by
Gigaset?
I use the N300IP. Supports 3 active SIP calls I
Does anyone know if Gigaset is for sale in the USA? Based on my assessment of
phones and features, i would like to try the N300IP base along with C610H
phones.
I can only find the handsets on ebay, no retailers in USA. And I suspect they
are using European frequencies.
--
That's how we do it - write to a memory based (ramdisk) disk then write to HDD
upon call completion. We haven't tried a SSD but that may be necessary
depending on your call volumes.
From: asterisk-users-boun...@lists.digium.com
I want to track the number of calls up at any given time, through the AMI. I
found the Link and Unlink commands as the most likely candidates - is that the
right way?
Also, a comment on the wiki suggests that Link may be called several times for
a single bridge if transcoding is required.
channels verbose”
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Thursday, October 18, 2012 9:58 AM
To: Asterisk Users List
Subject: [asterisk-users] Counting calls in progress from AMI
I want to track the number
take a look at AsteriskControl script at www.generationd.com
This is a free script that monitors, responds to IP address changes, etc. and
restarts asterisk.
You can also use HAAST (commercial) at same site - it can check for missing
registrations etc and restart asterisk too.
-=M=-
Check out smartCID on www.generationd.comhttp://www.generationd.com
This script allows lookup of incomming calls based on number and either Block
(no ring), endless ring (ignore), or pass through to asterisk. It allows
allows rewriting of CID name based on number. All numbers stored in a
...
For redundant/failover of Asterisk checkout HAAST at
www.generationd.comhttp://www.generationd.com The HAAST product sits between
Linux and Asterisk, monitors for failures etc, and then fails over to another
Asterisk box. It effectively creates a low-cost cluster, moving IP's etc to
Be careful with DRDB singe failing drive/corruption on one peers takes down the
other too...
Check out haast as well (at www.generationd.com) for a commercial asterisk
clustering solution.
Michelle
(GenerationD Systems)
From:
Is it possible to detect the failure of an agent to register with Asterisk via
the AMI ?
When I try to register with Asterisk 1.4 using an invalid password I don't see
any event in the AMI, but see this in the messages log:
[2013-10-05 22:05:03] NOTICE[24598] chan_sip.c: Registration from
Gareth:
Did you check if your message (or security) log recorded anything during these
attempts? If so, can you post the content of the logs during this attack?
M
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf
Is there a recent survey of that Linux distro and version people are using for
the Asterisk installations? I recall seeing a pie chart over a year ago (I
think on a wiki but I can't find it again)also hoping for something more
current.
I suspect RH5 and RH6 are most popular...but I'm
I need to disable/enable a peer after hours automatically, and am thinking
about doing so via the AMI.
Is there a command to enable/disable (or perhaps delete/add) a peer via the
AMI? I could create code to modify sip.conf and force a reload, but that seems
like the wrong approach...
--
someone tries to use it during the 'off' time. no
need for anything as brutal as disabling it in sip.conf.
On 2013-10-23 12:37 AM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
I need to disable/enable a peer after hours automatically, and am thinking
about doing so via the AMI
Is there a mapping of AMI versions to Asterisk versions somewhere? For
example, Asterisk 1.4 includes AMI version 1.0 (at least that's what I see when
I connect to Ast 1.4 via telnet to the AMI port)
Also, doe the AMI version changes reflect changes to the AMI commands? If so,
is there also
I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk
system has been stable for years, and has no trouble bridge SIP phone sets to
IAX trunks.
When I initiate a call from the IAX ATA, something goes wrong.One rare
occasion it works fine, but usually there is no
: [asterisk-users] IAX2 bridge failing
Michelle Dupuis wrote:
Some more details...I noticed that the call is bridged, and audio goes
one way. However, the dial command still times out after 35 seconds
(approx), and exists non-zero.
While the channels are up, I did an core show channel xxx
? Or something I can fix through config?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
[mdup...@ocg.ca]
Sent: Thursday, December 12, 2013 5:08 PM
To: Asterisk Users List
Subject: [asterisk-users] IAX2
Ok just restart
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Friday, December 13, 2013 11:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing
I tried
meant to say restart didn't help either..
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
[mdup...@ocg.ca]
Sent: Saturday, December 14, 2013 11:20 PM
To: Asterisk Users List
Subject: Re
: [asterisk-users] IAX2 bridge failing
Did you change your network switch recently? Some Digium IAX ATAs do not
behave well with Cisco equipment.
On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
meant to say restart didn't help either
Is there a mapping of AMI versions to Asterisk versions?
eg:
AMI 1.0 = Ast 1.4
AMI 1.1 = Ast 1.6
etc...
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
When I issue a 'core show channels' command I notice that long usernames (and
channel number) are truncated. For example, if the username is
FONEMITEL1234567890 for a trunk, then it will show
SIP
Privilege: Command
Channel Location State Application(Data)
Of Richard Mudgett
[rmudg...@digium.com]
Sent: Tuesday, January 21, 2014 6:12 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] core show channels truncates channel names?
On Tue, Jan 21, 2014 at 3:39 PM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
When I issue a 'core show
I'm looking at setting type=peer vs type=user (in both IAX and SIP conf
entries), and I found a comment attributed to digium
(http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that
type=user is depricated and that we should only use type=peer
Is that still correct? Will
I'm creating an AMI client and I only want to get newchannel events (as well as
responses to any actions I initiate). What would I set the eventmask to to
only get the newchannel events?
For anyone else looking...is there a table somewhere online that maps events to
their eventmask
Of Daniel Jenkins
[dan.jenkin...@gmail.com]
Sent: Thursday, January 23, 2014 9:03 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] AMI eventmask question
On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
Hi
I'm creating an AMI client and I
: [asterisk-users] AMI eventmask question
On Thu, Jan 23, 2014 at 3:06 PM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
That's an interesting link - I didn't know you could set a per user eventfilter
in the conf file
However, I'm hoping to do this in the AMI connection for more
Markus,
We are developing an Asterisk intrusion detection prevention tool which will
allow you to limit connections by geographic region
(continent/country/region/city), and include/exclude IP subnets, etc.
If you are interested let me know off-list (we're looking for beta testers!).
I'm looking for some beta testers to provide feedback on an Asterisk intrusion
detection prevention program we're releasing soon.
As a quick overview, the program provides:
- banning based on geographic location of source IP (Continent, country,
region, city, etc)
- detection and banning based
Some food for thought:
If you use DRBD, then you will mirror corruption from one system to another.
You also cannot selectively pick files in a folder to mirror (you will mirror a
lot!) As well, DRBD struggles as peers are set further apart (latency) or
number of changes increases.
A lot of
After each line of text, please also dip the corner of your keyboard into your
ink well to ensure your writing can been seen.
Calling something natural because it used to be that way isn't always correct.
-MD-
P.S. Notice how little we see PS in posts...now that we can also edit our own
I see a lot of attempts by hackers to call 00972595301123? or 011972595115207?
or variations but that same 972595 is often present.
Can someone break down that dial string with an explanation? The 011 look like
an overseas call (from Americas), while the 972595XX is unclear...
--
To: Asterisk Users List
Subject: Re: [asterisk-users] Numbers hackers call
On 26 Mar 2014, at 15:05, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
I see a lot of attempts by hackers to call 00972595301123? or 011972595115207?
or variations but that same 972595 is often present.
Can someone
I've noticed that the Asterisk (v11) security log captures attempts do dial
without first authenticating, and places the number dialed into the accountid
field.
I'm trying to distinguish between failed attempts to register and attempts to
dial without registering, but the security log treats
I (canadian) store has a deal on for the vera lite controller:
http://www.tigerdirect.ca/applications/searchtools/item-Details.asp?EdpNo=8930107sku=VEP-STARTER1
but this looks different than the vera lite green white:
?oops...wrong list :)
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of Michelle Dupuis
mdup...@ocg.ca
Sent: Friday, March 28, 2014 5:43 PM
To: Asterisk Users List
Subject: [asterisk-users] Best zwave controller
: Michelle Dupuis mdup...@ocg.ca
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Thursday, March 27, 2014 12:55:21 AM
Subject: [asterisk-users] Security log format / content
I've noticed that the Asterisk (v11) security log captures attempts
do dial without first authenticating
Take a look a SecAst from www.generationd.comhttp://www.generationd.com/
It does everything fail2ban does and more, including blocking users by
geography (we exclude all of Asia and Africa), detection of break-in patterns
(even if someone guessed your un/pw), detect changes in dial rates, etc.
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk 1.6
On Friday 04 Apr 2014, Michelle Dupuis wrote:
Take a look a SecAst from www.generationd.comhttp://www.generationd.com/
It does everything fail2ban does and more, including blocking users by
geography (we exclude all of Asia and Africa
If you know your users are all from with your country, or state, or even city,
you could restrict geographic access in your secast.conf file like this:
ruledefault=deny
ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA
The above would:
- By default deny all source IP's anywhere
IMHO: If you're announcing a product, selling a product, etc. it belongs on the
commercial list. If you're asking/answering questions about Asterisk and the
ecosystem I think you can mention commercial products too. (We don't want to
pretend they don't exist, and then steer users to only
These are at completely different levels of the ISO stack...question is making
sense to me.
(What does it mean to divert a call to a serial port). Do you mean route a
call over a link that is ppp/dialup and connected to another endpoint on the
other side of that link?
If so you would have to
Another alternative is SecAst (Asterisk intrusion detection system). Grab the
free version from www.generationd.comhttp://www.generationd.com/?
It does everything fail2ban does, plus you have the option of blocking IP's
based on geograhic origin, detecting suspicious call patterns, etc.
I have setup an Ast 11.6 host and I want to login via AJAM. I setup
manager.conf, http.conf described in the docs. When I login via the AMI it
works fine (see below), but when I login via AJAM the same credentials fail
(see further down)
Can someone tell me how to fix this?
---
: Friday, May 16, 2014 3:25 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
- Original Message -
From: Michelle Dupuis mdup...@ocg.ca
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Friday, May 16, 2014 2:43:30 PM
Subject: [asterisk
!
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of Michelle Dupuis
mdup...@ocg.ca
Sent: Friday, May 16, 2014 3:39 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
You're
PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
- Original Message -
From: Michelle Dupuis mdup...@ocg.ca
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Friday, May 16, 2014 3:39:35 PM
Subject: Re: [asterisk-users] Login by AMI ok
actually rawman and manager are very different, and you don't need cookies just
to test login. However, I found the problem: I forgot quotes around the curl
command.
Thanks!
--
_
-- Bandwidth and Colocation Provided by
After reading about the 2 major SSL (and TLS?) weaknesses discovered this
year, I was wondering how it affects asterisk.
Does the SIP authentication use TLS - or something that was recently broken?
Is there a risk of exposing passwords?
Thanks!
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If you have a small Asterisk installation install the free version of SecAst:
http://www.voip-info.org/wiki/view/SecAst+(Asterisk+Intrusion+Detection+and+Prevention)
For general Asterisk security info check this out:
http://www.voip-info.org/wiki/view/Asterisk+security
-=Michelle=-
All
You might get a better response on the FreePBX forum. (FreePBX adds pre-built
dialplan elements onto standard asterisk. This forum is more for Asterisk)
But some suggestions:
SSH to your PBX
enter the Asterisk CLI
set verbose to 10
Call into the problematic number
...and watch where the call
You can also take a look at SecAst (www.generationd.com).The free version
is a drop-in replacement for fail2ban but also add a lot more intelligence (and
no need to update regex's etc). There's also geographic IP fencing so you can
block attacks by country / region / city etc., only allow
There are lots of ways to solve this, and NOT to solve this. Don't start
adding lots of rules to iptables (or deep per packet inspection requirements)
as this will hurt capacity...and it doesn't really solve the problem
Take a look at
http://www.voip-info.org/wiki/view/Asterisk+security
If
When I issue the CLI command 'core show calls' I see how many calls have been
processed by Asterisk since it started; eg:
0 active calls
198 calls processed
Is there a way to reset the calls processed counter without having to shutdown
and restart asterisk?
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