I see asterisk is finding res_jabber.so not compiled for your asterisk
version. As Tim just said, remove all the modules from
/usr/lib/asterisk/modules and reinstall asterisk.
[2013-01-10 14:20:10] WARNING[27062]: loader.c:804 inspect_module: Module
'res_jabber.so' was not compiled with the same
Hello,
I am playing with the manager interface and it seems I cannot catch the
event of a phone subscribing to an hint. Is there a way to catch this kind
of event using the manager interface? I use custom device states, so when a
phone subscribe to a hint, the device is created on the fly. I'd
2013/1/8 Lenz Emilitri lenz.lo...@gmail.com
2013/1/5 joachim zoach...@securax.org
You are pretty much limited to measuring the delay and the jitter.
The delay you can somewhat estimate prior to the call (with qualify for
example).
The jitter / packetloss you can only figure out when the
2013/1/8 Luis H. Forchesatto luisforchesa...@gmail.com
Greetings.
I got two extensions on my asterisk that autenticates from outside our
network, via internet. Is there a way to monitor, in certain time periods,
if they are available (online) and send some sort of notification if they
H. Forchesatto luisforchesa...@gmail.com
Hmmmlooks good, but I'm looking for something that I could do.
I'm not much of outsorcing.
2013/1/8 Leandro Dardini ldard...@gmail.com
2013/1/8 Luis H. Forchesatto luisforchesa...@gmail.com
Greetings.
I got two extensions on my asterisk
2013/1/5 XBrian bobo...@yahoo.co.uk
Can I restrictthe number of concurrent registrations per friend?
Your question has no meaning. The registration is the way a peer says to
asterisk which is the IP address and port to use to contact him. There can
be just one registration active at time. If
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
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
I don't know how many I/O can be achieved on a modern hardware, but I don't
think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of
data. However can be a good idea to start loading a server and be prepared
to share the load on another server.
Leandro
2013/1/2 Steve Totaro
You should start getting a PRI card. I have good result with both Sangoma
and Digium one. After having configured the card in the system (libpri,
dahdi and asterisk part), it is a matter of few asterisk configuration row
to save all calls to a wav file.
For example, if your incoming calls are put
Have you configured the canreinvite=yes in sip peer?
I am currently off work for two days, but a 100% fail means a configuration
problem for sure.
Leandro
2012/12/27 Eric Wieling ewiel...@nyigc.com
We are offering $100 (paid via paypal or check) to the first person who
assists us in
I usually set directmedia=yes with good results...
Leandro
2012/12/27 Christopher Harrington ch...@acsdi.com
On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling ewiel...@nyigc.com wrote:
sip.conf settings:
directmedia=yes
I know you've said you tried it both ways, but consensus seems to be
If you are using an analogue/sip ata, then the problem is on the ata. Run a
packet capture and you'll see the invite coming from the ata without nobody
using the phone...
I am typing from my mobile phone...
Il giorno 11/dic/2012 18:55, Joseph syscon...@gmail.com ha scritto:
On 12/11/12 11:48,
** **
Maybe,
** **
You can do that, with queues, and ringall strategy.
On Wed, Dec 5, 2012 at 4:53 PM, Leandro Dardini ldard...@gmail.com
wrote:
You can dial all the extensions at once, putting all them in the dial
string, separated by . There is no other method
Yes, go for it. However I have added another autoincrement column and
created the primary key on it. On the other columns I need to search I have
created just an index.
Leandro
2012/12/6 Olivier oza_4...@yahoo.fr
Hello,
I need to develop an application that will query (mostly reading) an
ask about the impact of making an existing column a
primary key, in a MySQL forum.
Leandro's solution seems to be a good one as well and does guarantee
uniqueness.
Ron
On 06/12/2012 12:25 PM, Leandro Dardini wrote:
Yes, go for it. However I have added another autoincrement column
You can dial all the extensions at once, putting all them in the dial
string, separated by . There is no other method.
Leandro
2012/12/5 Paolo De Michele pa...@paolodemichele.it
hi all,
I want have an information about ring group in asterisk (1.8.16 - centos
6.3)
I have configured
I suspect you have something wrong in your server hardware... have you
tried running a memtest?
Leandro
2012/11/27 Adolphus Enaboifo adolphus.enabo...@osenkorp.com
Hi List members,
Thanks for the support so far as I try to install and test my first
asterisk system.
I was able to finally
I am not really sure, restarting asterisk and dahdi can be the most obvious
thing to do, but restarting the dahdi kernel module can be useless if you
haven't changed the kernel module configuration and reloading the module in
asterisk can be enough if you have changed just the chan_dahdi.conf
2012/11/20 gincantalupo gincantal...@fgasoftware.com
Hi all,
I have problems dialling out because my new telco (the previous gave no
problems) tells me my PBX adds a leading 0 and that's why I cannot dial out
(but I can receive calls).
I make a small extensions.conf as a test:
exten =
...
Moreover we had no problem with the previous telco (fastweb).
So we can only call PTSN numbersnot mobile phones.
Giorgio
On 11/20/2012 11:12 AM, Leandro Dardini wrote:
2012/11/20 gincantalupo gincantal...@fgasoftware.com
Hi all,
I have problems dialling out because my new telco
Not only, you have to restart dahdi/zaptel as well.
Leandro
2012/11/20 Frederic Van Espen frederic...@gmail.com
On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote:
I'm sure nobody has added something... tried prilocaldialplan and
pridialplan but nothing changed.
Question: if
Hello,
I want to manage hints in a different way, putting all the hints in the
same context and trying to recognize the subscribing peer, but I can't find
any variable set about the calling peer. Peers need to be authenticated to
be able to subscribe to the hint, but I am not able to access any of
Thank you,
I think I'll surrender in trying to use the realtime extension and use
instead the simple ODBC interface. However I'd like to access some channel
variables. Which ones are available inside the extension hint porcessing? I
tried ${CDR(accountcode)} and it is not available, nor the
Thank you again,
The problem in this setup is the inability to isolate a group of extensions
from others. I mean, if all hints are in the same context, each extension
can subscribe to any of the hints. The reason I prefer a completely
realtime hints was I'd like to dynamically create hints in
Hello,
I'd like to start using realtime hints in my asterisk 1.8 dialplan, but I
am unable. I haven't understood if they have to be put inside the
extensions realtime table (with priority -1) or if a dedicated realtime
hints table can be made. Neither ways seem to work. Have you any working
You can see if asterisk has been restarted by checking the number of calls
processed. If almost zero, it has been restarted.
core show calls
Leandro
2012/8/19 Jan Blom jan.b...@peopleinteractive.se
Hello,
** **
Is there a way to detect, via cli or any other way, that Asterisk is in
One of my clients uses thirdlane. The web interface is clean and nice, but
asterisk completely locks when one of the client change the config and
reloads during peak hours. It is possible my client uses an old version or
hasn't applied all the patches or hasn't configured asterisk in the right
way
Sure, you can include multiple files from the general extension.conf. You
can do the same for the sip.conf.
Leandro
I am typing from my mobile phone...
Il giorno 11/ago/2012 12:17, Kannan vasdevelo...@gmail.com ha scritto:
Hi List,
I am planning a multi-tenant VoIP services system with
2012/8/11 Hatos Gabor ha...@ggki.hu
Hi Team,
I use Asterix 1.6.2.9-2 what is running on debian squeeze. I completely
statisfied this software. I did everything I want so far. I love it so
much, but there is a point where I can not step through.
1)
I have connected to my telephone
2012/8/9 Ashish Agarwal ashisha...@gmail.com
Hello,
I have two asterisk boxes running and both are using DAHDI PRI Card. I
wish to know if IAX is the best method to connect both the boxes?
IAX2 is a great protocol, it can do amazing things in saving bandwidth
(with the trunking feature) and
Let us know how does it performs...
Leandro
2012/8/6 Shahid H shah...@gmail.com
I have bought a new server today:
i7-2600 CPU, 8GB and 2 x 256GB SSDs. 100Mbit Connection.
I hope CPU is powerful enough for 200 concurrent calls.
On Sun, Aug 5, 2012 at 1:57 AM, Michelle Dupuis
The busiest server I am managing reaches 120 concurrent channels (with
mixed recording). It is a dual processor, dual core Intel 5150 with 16 GB
of ram and raid sas controller. The load reaches rarely 3.0.
Having to double the number of channels and due to the 100% call
recordings, I'll go with a
Raid 10 (4 hard drives)?
On Sat, Aug 4, 2012 at 10:17 AM, Leandro Dardini ldard...@gmail.comwrote:
The busiest server I am managing reaches 120 concurrent channels (with
mixed recording). It is a dual processor, dual core Intel 5150 with 16 GB
of ram and raid sas controller. The load reaches
PM, Leandro Dardini ldard...@gmail.comwrote:
It is not necessary to use an high performance drive. The bottleneck will
be the processor, not the disk. A single disk can handle ten times the load
of 200 ulaw channels.
Leandro
Il giorno 04/ago/2012 12:39, Shahid H shah...@gmail.com ha scritto
to SATA
HDD.
What do you think of this?
On Sat, Aug 4, 2012 at 5:51 PM, Benny Amorsen benny+use...@amorsen.dkwrote:
Leandro Dardini ldard...@gmail.com writes:
A single sata disk will be an unacceptable single point of failure. Get
three disks and get in raid5 configuration. You'll gain
It is reasonable 'n' is not usable as priority number. How can asterisk
know the second priority if all other priority have 'n' as priority number?
In a relational database there is no 'sequential read'.
In other words, you need to assign the priority to all entries.
Leandro
Il giorno
If you check the contrib/realtime direco
2012/8/3 Daniel-Constantin Mierla mico...@gmail.com
Hello,
I was wondering if there is a tool that can create the realtime database
structure for latest Asterisk version or a web resource/file containing the
sql scripts. Hope I haven't missed obvious
If you check the contrib/realtime/mysql directory in the source tree,
you'll find scripts for almost all the tables.
Leandro
2012/8/3 Daniel-Constantin Mierla mico...@gmail.com
Hello,
I was wondering if there is a tool that can create the realtime database
structure for latest Asterisk
was
written. but know it does so what do we do. Unfortunately I am not a ansi C
guy or I could probably fix it .
Thanks
Bryant Zimmerman (ZK Tech Inc.)
--
*From*: Leandro Dardini ldard...@gmail.com
*Sent*: Friday, August 03, 2012 2:18 AM
*To*: Asterisk Users Mailing List
No, numbers have to be in sequence.
Leandro
I am typing from my mobile phone...
Il giorno 03/ago/2012 20:28, Raj Mathur (राज माथुर) r...@linux-delhi.org
ha scritto:
On Friday 03 Aug 2012, C. Savinovich wrote:
You don't use 'n's in your dialplan?, you number it yourself?
man, what if
asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] Multi-Tenant PBX with Asterisk
Thanks Leandro for your comments.
On Mon, Jul 30, 2012 at 6:35 PM, Leandro Dardini ldard...@gmail.comwrote:
2012/7/30 Kannan vasdevelo...@gmail.com
Hi
I came across couple of pointers
2012/7/30 Kannan vasdevelo...@gmail.com
Hi
I came across couple of pointers on the Internet regarding solutions
available for providing hosted PBX service.
1. Multiple PBXs: Using separate hardware to host each PBX. Pretty
straightforward, but no hosting company wants to use it.
2.
ARA is an acronym for Asterisk Realtime Architecture and is a different way
to keep configuration files in asterisk. Instead of reading configuration
from plain files at startup, asterisk read them from database, in realtime.
This mean, if you need to add a peer, you drop a new line in the
Asterisk has some configuration files, like sip.conf holding the peers and
trunks details and the most important one, available in several flavours,
extensions.conf, extensions.ael... these latest ones are merged toghether
at run time. The extension conf file is referred also as 'dialplan' and it
Hello friends,
I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can
access the caller Call ID (fbasename field in voipmonitor cdr table)
looking at the SIPCALLID variable in asterisk, but how can I access from
within asterisk the Call ID of the second leg of the call (the one
Hello friends,
I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can
access the caller Call ID (fbasename field in voipmonitor cdr table)
looking at the SIPCALLID variable in asterisk, but how can I access from
within asterisk the Call ID of the second leg of the call (the one
Hello friends,
I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can
access the caller Call ID (fbasename field in voipmonitor cdr) looking at
the SIPCALLID variable in asterisk, but how can I access from within
asterisk the Call ID of the second leg of the call (the one
15k users are quite a big number. To my clients with a large user base I
advice always to partition the load on multiple servers. This has a list of
advantages, like the ability to power cycle a node without impacting all
your users, easier debug and tests of problems and solutions, abiity to
Answers in text.
2012/7/23 Kannan vasdevelo...@gmail.com
Thanks Leandro for your reply. See my comments inline.
On Mon, Jul 23, 2012 at 12:57 PM, Leandro Dardini ldard...@gmail.comwrote:
15k users are quite a big number. To my clients with a large user base I
advice always to partition
2012/7/13 Nikolay G. Petrov r...@dir.bg
Hi guys!
I have a some non standard problem when I register my asterisk into My
SIP Provider .
The trouble is: my asterisk stay behind router with port forwarding, who
have Public IP (55.55.55.55 - for example), asterisk have a private IP
Il giorno 13/lug/2012 14:00, Nikolay G. Petrov r...@mail.bg ha scritto:
13.07.2012 15:01, Leandro Dardini пишет:
2012/7/13 Nikolay G. Petrov r...@dir.bg
Hi guys!
I have a some non standard problem when I register my asterisk into My
SIP Provider .
The trouble is: my asterisk stay behind
Port 5060 when used with the sip protocol is used witj UDP protocol. Telnet
is using TCP.
I am typing from my mobile phone...
Il giorno 01/lug/2012 09:35, alok srivastava alok...@gmail.com ha
scritto:
dear
i have configured properly asterisk. At the one end i am using x-lite soft
ph and
20.000 users is really a big number, as big as 2000 concurrent calls.
As previously stated on this list, it depends... it depends by the type of
calls for example. If all media is offloaded from the server letting the
phones to reinvite each other, than your server CAN support the call
volume. If
Hello,
I am not able to jump to a label from inside a macro. The goto is made
inside a catch while the label is in the body of the macro:
macro recordMessage() {
Answer();
recordagain:
Playback(after-the-tone);
Playback(say-temp-msg-prs-pound);
record(/tmp/${UNIQUEID}.wav);
,DIALTIMEOUT)=${ODBC_GET_HUNTLIST_TYPE(${ID})});
Leandro
2012/5/5 Jonas Kellens jonas.kell...@telenet.be
**
Will ODBC become the default then ?
I see no ODBC-command to use in the dialplan.
Jonas.
On 05/05/2012 11:12 AM, Leandro Dardini wrote:
Use ODBC. Check the func_odbc.conf
Use ODBC. Check the func_odbc.conf configuration file.
Leandro
2012/5/5 Jonas Kellens jonas.kell...@telenet.be
**
Hello,
I notice when upgrading from 1.6.2 to 1.8 that in the menuselect
app_mysql is indicated as deprecated.
If one wants to use the MySQL-command in the dialplan, how to do
2012/5/2 Kamlesh Kumar kamlesh_...@hotmail.com
Hi,
I need to configure global parameters in sip.conf like rtptimeout,
rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time
architecture. Please suggest the way to do it.
thanks,
Kamlesh
For what I have discovered, it
Hello,
I was tired of manually aligning ael files in emacs, so I downloaded the
.el file on
http://www.voip-info.org/wiki/view/EMacs+Asterisk+Syntax+Highlighting
Unfortunately there is a problem with switch statement. Do you know of a
better .el file or are you good in writing .el files to fix
Check the command Busy() of the dialplan, it return the busy state at the
calling party.
Leandro
2012/4/26 Jonas Kellens jonas.kell...@telenet.be
**
Hello,
can someone please tell me if this is possible and how ?
Kind regards,
Jonas.
On 04/24/2012 12:59 PM, Jonas Kellens wrote:
2012/4/25 Olivier CALVANO o.calv...@gmail.com
Sure, sorry for the Confusion ;=)
Server A Trader:
Asterisk Server 1.6.x for call routing only.
IP Adress: 172.16.0.11
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Server B
1. No, asterisk can act as pbx and as conference server
2. No, just bought a powerful server
3. Not me, sorry
4. You are limited only by the CPU of your server
Il giorno 20/apr/2012 19:21, Mitchell Johnson mitch.johns...@gmail.com
ha scritto:
We're looking into using Asterisk to do our
I used asterisk with some dialplan customization. It is not difficult. All
client asterisk step in the central asterisk to reach the providers. I have
a central system to monitor calls, call quality, enforce limits and route
versus the best provider. To be sure I have two asterisk servers in
Your understanding of the problem seems incorrect. The problem seems due to
the extension not available in your dialplan. Please check carefully in
which context the call is placed and if the extension is defined in that
context.
Maybe it can be useful to define a _X. extension to catch all not
Your problem originate from the use of insecure option. Using this option,
the peer is authenticated using the registration ip and not the user and
password.
Leandro
Il giorno 26/mar/2012 05:48, YeungJoe ma_ch1...@hotmail.com ha scritto:
Hello All,
I am Asterisk user, and right now I have
Hello,
I have a problem with premature media and inband progress audio. I am using
the latest 1.8.10.1 and this is the setup:
soft phone --- asterisk --- SIP provider
The number I call is giving back some hints via inband audio I am not able
to ear from the soft phone. They stop on the asterisk
and find out where that early media is going.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
Leandro Dardini ldard...@gmail.com wrote:
Hello,
I have
30030
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
Leandro Dardini ldard...@gmail.com wrote:
All NAT and firewall problems are already been excluded. All peers are on
public IP address and no firewall is active between them. The missing
routing of the audio
Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
Leandro Dardini ldard...@gmail.com wrote:
The asterisk box has only one interface. I am capturing all the traffic on
the box and the only audio traffic is from the provider
Bingo, it was the r option!
Thank you
Leandro
2012/3/25 isr...@gmail.com
Do you have r in your dial string?
If yes remove that
-Original Message-
From: Leandro Dardini ldard...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Sun, 25 Mar 2012 11:35:45
To: Asterisk
If you have 10 billing plans from different providers, you have for sure at
least almost all the data. Use the prefix from the plans to build your own
database of prefixes and destinations.
Il giorno 23/mar/2012 05:06, Mikhail Lischuk mlisc...@itx.com.ua ha
scritto:
**
Is it a problem to parse
2012/3/22 Zohair Raza engineerzuhairr...@gmail.com
Hi,
How to allow registered sip users to call without re-authentication
insecure =yes/very are deprecated in 1.8
I want to avoid fromuser= in peer configuration. When I add this in peer
asterisk, my asterisk accepts call otherwise it says
speed, not
changing with the call load!
On Fri, Mar 2, 2012 at 12:20 PM, Leandro Dardini ldard...@gmail.comwrote:
Asterisk can cache cdr records to avoid having to write continuosly in
the cdr backend. Writing in bunch instead one at once improves performance.
Check the cdr.conf file and disable
Asterisk can cache cdr records to avoid having to write continuosly in the
cdr backend. Writing in bunch instead one at once improves performance.
Check the cdr.conf file and disable the option batch if it hurts you.
Leandro
Il giorno 02/mar/2012 07:24, [Digital^Dude] ® millennium@gmail.com
I prefer multiple servers sharing the load. All asterisk based. This
let me scale up the power of the system just adding more servers. I
use asterisk 1.8 realtime with all the data (peers, voicemails, ivr
messages and so on) stored in a pair of mysql database with
multimaster replication. Phones
2012/2/10 Brynjolfur Thorvardsson bi...@itanet.nu:
I hope I'm not flogging a dead horse here, but the discussion around the
whole scalability issue in Asterisk have opened my eyes to a whole lot of
issues, making me increasingly confused!
We have a fully functioning and stable installation
/10/12, Leandro Dardini ldard...@gmail.com wrote:
mysql multimaster replication and
asterisk realtime.
Just a word of caution: I've had terrible luck with MySQL NDB tables in a
multimaster setup. I'm not a big expert but v.5.0 and 5.1 have given me
lots of reliability issues (I lost table
I can assure you it works. It is important you can set in the [general] section:
match_auth_username=yes
Leandro
2012/1/19 Frank Church voi...@gmail.com:
Does Asterisk permit multiple registrations to the same host?
Each registration has a different username and password
The purpose is for
Me too, an maybe other people on the list are interested in knowing
your effort result and maybe appreciate a guide on the topic.
Thank you
Leandro
2012/1/13 Ronald Cepres rbcep...@gmail.com:
Hi Ishfaq,
Thanks for your reply. I've already started trying the XMPP method so I
can't help you
2011/12/27 virendra bhati virbh...@gmail.com
Hi list someone is trying to hack my server . Is there any way by whcih I
can stop hacking of my server except iptables ? I want to stop on the basis
of sip.conf account only. bcoz I can't apply iptables rules on server it's
remote server of server
Yes, this is one of my entries:
[trunk1]
context=fromoutside
type=friend
deny=0.0.0.0/0.0.0.0
permit=34.2.10.24
qualify=yes
2011/12/27 virendra bhati virbh...@gmail.com
Can you give an example how to set these oprion ...
On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini ldard
type=friend
http://0.0.0.0/0.0.0.0
permit=34.2.10.24
qualify=yes
So will it be fine or not ? Or it will get rest information from sip.conf
general section ?
On Tue, Dec 27, 2011 at 2:21 PM, Leandro Dardini ldard...@gmail.comwrote:
Yes, this is one of my entries:
[trunk1]
context
From time to time a similar subject pops up on the list. I'd like to repeat
it is extremely dangerous to ban IP based on a suspicious UDP activity. The
source IP of an UDP packet can be easily forged, so if you start using
fail2ban or other blacklist techniques, it can be very awesome to start
Add match_auth_username=yes in the [general] section of sip.conf
Remove from each peer any insecure entry
Usually I add also auth, defaultuser and username to the peer
definition, but some of these entries are not needed.
Leandro
2011/9/23 David Björkevik da...@dynamore.se
Dear list,
We
tried this and it's still the same.
(although I still have _unrelated_ peers with the insecure entry)
/David
On 2011-09-23 14:24, Leandro Dardini wrote:
Add match_auth_username=yes in the [general] section of sip.conf
Remove from each peer any insecure entry
Usually I add also auth
2011/9/5 Catalin S. jonsonpla...@gmail.com
Hello,
I have a problem with some variables in 1.8.6.0. I set on extension the
following lines:
exten = h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio,
local_lostpackets)}) ; lost packets by local end **
exten = h, n, Set (CDR (PCR) =
Hello,
Asterisk seems to try to authenticate incoming INVITE based on the [section]
in sip.conf and not the username specified.
I just removed the insecure option from my sip.conf requesting every
connection to be authenticated. I added the match_auth_username=yes in the
[general] section for
Hello,
for billing purpose between a multitenant asterisk box and another asterisk,
I am in the need to maintain multiple SIP trunks between them. Usually I use
insecure=invite,port but I had to remove or the trunks will be selected
based on IP address and not with username/password. I had to use
2011/5/15 RSCL Mumbai rscl.mum...@gmail.com
On Sat, May 14, 2011 at 11:43 AM, Leandro Dardini ldard...@gmail.comwrote:
Check if someone is brute forcing your asterisk accounts. It used to
happen to me before I install fail2ban. You can easily check the full log
of asterisk or with just
2011/5/14 RSCL Mumbai rscl.mum...@gmail.com
Hi,
On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest.
Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly
1-2 concurrent calls. No other activity on server. Top shows asterisk on
top.
Its quad xeon
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