Anyone?
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On 05/24/2015 11:01 PM, Mehdi Shirazi wrote:
Hi
I want to load balance SIP calls between two(or more)
Asterisks with only DNS SRV. I used bidirectional sync
Unison to synchronize configuration files and internal database file
between two Asterisk boxes.
The problem is when a calls come to
I came accross this article (Asterisk rtp mprovements
http://www.voip-forum.com/opensource/2013-04/asterisk-rtp-improvements/)
mentioning DNS based load balancing.
I will give Opensips loadbalance module further reading to better
understand how it works
Thanks for the tip.
2013/4/25
You the couple opensips + asterisk will help you. Opensips loadbalance module
is your friend.
Sent from my iPhone
On Apr 25, 2013, at 11:44 AM, Olivier oza_4...@yahoo.fr wrote:
Hello,
I've been given the task to study what would a good way to load balance SIP
trafic.
The prospective
Foundry serverIron does support SIP and its ASIC not a linux box Load
balancer like F5,
Refer to Chapter 10 (page 677) of ServerIron manual.
It explains everything in detail.
Also you may need to play with source nat a little bit to make your specific
configuration work, but it should work, at
2008/11/20 Nitzan Kon [EMAIL PROTECTED]
Hello!
We're looking for a solution to reliably load balance our
Asterisk boxes. So far we've been using a hodge-podge of
directing different services to different boxes/IPs, but
eventually I'd like to consolidate things so we can present
a single IP
--- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote:
2 openser servers with 3 ip adresses (1 virtual) +
heartbeat to ensure the
failover + watchdog to ensure if opensips/kamalio/openser
crashes a nice
failover reboot, it is working stable here
(dispatching to 10 servers +
What do you mean by hardware options? There are no ASIC-assisted SIP
load balancers out there. :-) The embedded hardware-based options
are load balancers built just like PCs - often on top of a UNIX kernel -
that run a software application-aware load balancing suite.
Your best bet is a
Hardware solutions are of course simply packaged software solutions.
Personally I would go with something that has this wonderful support base
and quick solutions versus dealing with a vendor. You did mention that
price was a consideration, right?
j
On Thu, 20 Nov 2008, Nitzan Kon wrote:
Alex,
I realize and agree that hardware load balancers are actually
software based. I'm less concerned about that and more about the
general specs:
Foundry ServerIron XL: rated for 1,000,000 concurrent connections
Linux box where OpenSIPS is sitting: rated for ...???
Not to mention a simple
Nitzan Kon wrote:
Foundry ServerIron XL: rated for 1,000,000 concurrent connections
Linux box where OpenSIPS is sitting: rated for ...???
Because OpenSER's load balancer is hash-based and not stateful, it is
rated for far, far more than that.
--
Alex Balashov
Evariste Systems
Web:
Unless the LB is SIP-aware, and can maintain a SIP session, I don't see
how it would work. As the SIP command stream sends discrete commands,
without some sort of basic level of session awareness, there's no
guarantee over a reasonable-length call that the INVITE and BYE would
even get sent to the
The solution to make this work and still work statelessly is to hash
various unique identifying bits of the SIP headers without maintaining
transactional, session or dialog information as such.
SIP wrote:
Unless the LB is SIP-aware, and can maintain a SIP session, I don't see
how it would
This baby talks about being able to do hardware SIP load balancing.
http://www.f5.com/news-press-events/press/2007/20070212.html
I've never used an f5 product so I can't provide any comments from
experience. I did look at an f5 load balancer product once and the
price was over 6 figures that was
N,
SIP-aware LBs do exist - but way way out of my price range.
Alex,
Remember we are an Asterisk-based provider. I'm not going
to drop enough money on a load balancer to go bankrupt. ;) That's
exactly why I'm wondering if it's possible to do this with a
DUMB load balancer. i.e. one that would
I was about to say, I'm sure F5 can do it... but...
price was over 6 figures
Why??!
It's spending money on these types of things when they are unnecessary
that is the undoing of every struggling VoIP provider I watch, in the
misguided belief that only will half a million dollars get you
Nitzan Kon wrote:
My concerns with OpenSIPS:
1. It's a software based solution, which means higher chance
of software-related failure, and higher chance of failure due
to problems with the Linux box hosting it.
A little bit of proper engineering will overcome that reasonably.
2. Overkill
Alex Balashov wrote:
I was about to say, I'm sure F5 can do it... but...
price was over 6 figures
Why??!
It's spending money on these types of things when they are unnecessary
that is the undoing of every struggling VoIP provider I watch, in the
misguided belief that only will half a
2. Overkill to install and maintain (if we can get a simpler
solution)
I am not agreed on point 2:
If I understood how to install opensips + heartbeat WITHOUT knowing any
program (opensips ? heartbear ?) or programming language(hell yes!) in a
week ( just knew what's invite and bye ;) a more
--- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote:
I am not agreed on point 2:
If I understood how to install opensips + heartbeat WITHOUT
knowing any
program (opensips ? heartbear ?) or programming
language(hell yes!) in a
week ( just knew what's invite and bye ;) a more
Nitzan Kon wrote:
--- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote:
I am not agreed on point 2:
If I understood how to install opensips + heartbeat WITHOUT
knowing any
program (opensips ? heartbear ?) or programming
language(hell yes!) in a
week ( just knew what's
3. Incoming calls - I admit complete ignorance. I don't know
how OpenSIPS handles incoming calls, but for those to arrive
at the user reliably they must arrive from the same IP address
the user is registered to. Otherwise their broadband router's
NAT firewall will just block the connection.
SIP wrote:
As for the current F5 SIP load balancer, we tried it a few years back
and it was a dismal failure. It wanted to do cookie-based SIP load
balancing and only worked with certain SIP proxies.
I assume that is because there is no way RFC-supported way to insert a
cookie into a SIP
For outbound trunking we go directly from Asterisk to the terminating
gateway no SIP Proxy involved. For inbound trunking we do go through
the SIP Proxy for the same reason you get users to. Incoming calls are
going to be more reliable if they are not tied to a single Asterisk
server (I
There are a few gotchas with a SIP Proxy the main one being transfers.
But if you can get away with not allowing transfers then you are best
to do so as the CDR's Asterisk produces are wrong anyway.
What is the transfer problem? Is it the Asterisk native type using
features.conf or the SIP
On Fri, Feb 29, 2008 at 2:01 AM, Ron [EMAIL PROTECTED] wrote:
Hi All,
If i have this kind of setup, what do i need to make it's load balance.
[ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
| | | |
Hi Greyman,
Should it look like this now? Can i use 2 SIP Proxies just to make sure
i have a backup? will that cause any problem again with regards to
calling extension to extension? Extensions will register on the asterisk
still? How about outbound calls to other SIP provider or a
If i have this kind of setup, what do i need to make it's load balance.
[ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
| | | |
-
|
On Fri, Feb 29, 2008 at 4:03 AM, Ron [EMAIL PROTECTED] wrote:
Hi Greyman,
Should it look like this now? Can i use 2 SIP Proxies just to make sure
i have a backup? will that cause any problem again with regards to
calling extension to extension? Extensions will register on the asterisk
On Fri, 29 Feb 2008 6:21 +0200, Yehavi Bourvine +972-8-9489444
[EMAIL PROTECTED] wrote:
See the discussion a few days ago. The Asterisk server saves the value of
SYSNAME (defined in asterisk.conf) in the field REGSERVER inside MySQL.
Regards, __Yehavi:
Ahh
Hello,
Here is how I do this. The prerequisits are:
- MySQL to hold the extensions realtime database. MySQL is synchronized
among all servers using the Master/slave replication model.
- The phones are spread by some external algorithm over the Asterisk servers
(statefull load balancer,
Vieri wrote:
What I would like to do is have two identical *
servers which accept registrations of sip extensions
4000-4999.
If I define a rrDNS or LinuxHA then I should have
load-balanced registrations.
However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001
On Fri, 22 Feb 2008 19:44 +0200, Yehavi Bourvine +972-8-9489444
[EMAIL PROTECTED] wrote:
When a call arrives I check whether the REGSERVER coloumn is the same as
the
local server or not. If not, then there are two options:
- Pass the call via IAX to the other servers; this makes both server
--- Anthony Francis [EMAIL PROTECTED] wrote:
Have you tried placing the sip registrations in a db
using realtime?
I'm not that sure I want to use realtime because I
would then depend on the sql service never failing (I
could use clustered active-active MySQL but that
sounds overkill, or maybe
On Fri, Feb 22, 2008 at 11:42 AM, Vieri [EMAIL PROTECTED] wrote:
However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001 tries to call 4002 then I
would like to do something like:
- lookup 4002 on *1, try to establish a call if it's
REGISTERED here
- if it's
What I would like to do is have two identical *
servers which accept registrations of sip extensions
4000-4999.
If I define a rrDNS or LinuxHA then I should have
load-balanced registrations.
However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001 tries to call
--- Andres Jimenez [EMAIL PROTECTED] wrote:
On Fri, Feb 22, 2008 at 11:42 AM, Vieri
[EMAIL PROTECTED] wrote:
However, say ext. 4001 is registered on *1 and
4002 is
registered on *2, if 4001 tries to call 4002 then
I
would like to do something like:
- lookup 4002 on *1, try to
On Fri, Feb 22, 2008 at 5:49 PM, Vieri [EMAIL PROTECTED] wrote:
--- Andres Jimenez [EMAIL PROTECTED] wrote:
On Fri, Feb 22, 2008 at 11:42 AM, Vieri
[EMAIL PROTECTED] wrote:
However, say ext. 4001 is registered on *1 and
4002 is
registered on *2, if 4001 tries to call 4002
On Fri, Feb 22, 2008 at 5:49 PM, Vieri [EMAIL PROTECTED] wrote:
Thanks. I'll try that although I hope it won't go into
an infinite loop between the 2 servers.
You are right. That could happen if the phone is not registered anywhere
You can put some security in the dialplan.
if calls
I tried to use DUNDi on my local servers but I can't
seem to make it work. Most howtos out there explain
the use of DUNDi when the extension ranges do not
overlap.
The following doc describes using the same extensions across multiple *
servers. It requires using realtime, but seems to do
: Wednesday, December 12, 2007 11:28 AM
Subject: Re: [asterisk-users] Load Balancing over 2 E1 Lines
On Dec 12, 2007 8:08 AM, Eric Delaporte [EMAIL PROTECTED] wrote:
I read something about DIAL(Zap/r1/…) for using round robin, and it seems
to
work.
That will give you the same number of calls routed
On Dec 12, 2007 8:08 AM, Eric Delaporte [EMAIL PROTECTED] wrote:
I read something about DIAL(Zap/r1/…) for using round robin, and it seems to
work.
That will give you the same number of calls routed to each line
Is there any other possible way to make sure that all lines are used in the
Why not Random application available in Asterisk ?
quite simple I believe.
asterisk1*CLI show application Random
-= Info about application 'Random' =-
[Synopsis]
Conditionally branches, based upon a probability
[Description]
Random([probability]:[[context|]extension|]priority)
probability
On Wed, 15 Aug 2007, Nicholas Blasgen wrote:
I have 10 SIP trunks that I'd really like to round-robin load balance.
Currently I have a macro that switches between available lines, but there
really must be a function in Asterisk to do this on its own. So my question
is just that, are there
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nicholas Blasgen wrote:
I have 10 SIP trunks that I'd really like to round-robin load balance.
Currently I have a macro that switches between available lines, but there
really must be a function in Asterisk to do this on its own. So my question
Hi,
Kamran Ahmad wrote:
any idea how to loadbalance IAX2 trafic to multiple
asteirsk
Use app_random:
exten = _X.,2,Random(50:6)
exten = _X.,3,Dial(IAX2/server01/${EXTEN})
exten = _X.,4,Dial(IAX2/server02/${EXTEN})
exten = _X.,5,Goto(8)
exten = _X.,6,Dial(IAX2/server02/${EXTEN})
exten =
Yves wrote:
Hello,
I'm trying to find a good solution for load-balancing of several
Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have
a front VoIP router (or several) who dispatches the calls of the
different boxes.
This routing can be done with SER for SIP (redirect
On 00:03, Mon 13 Jun 05, Matt Riddell wrote:
Yves wrote:
Hello,
I'm trying to find a good solution for load-balancing of several
Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have
a front VoIP router (or several) who dispatches the calls of the
different boxes.
I was more thinking about a linux-ha heartbeat system between the
redundant devices. The voicemail propagation is not a problem as I don't
use it, and the dialplan is stored on a remote database (that's another
problem to make redundant :) ).
The point I have to check now is how to configure
I need to do load balancing only for the following functionalities:
1) Registration of SIP clients to * servers.
2) Load balancing of the INVITEs from SIP clients to different * servers.
I'm not interested in supporting the features, which you have
mentioned below. I'm not aware how the
Jagan Mohan wrote:
I need to do load balancing only for the following functionalities:
1) Registration of SIP clients to * servers.
2) Load balancing of the INVITEs from SIP clients to different *
servers.
I'm not interested in supporting the features, which you have
mentioned below.
How do you plan on supporting call queues, parking and agents with 2 *
servers? This is something that has blocked us from being able to do our own
SER-based load balancing.
-Matthew
Jagan Mohan wrote:
Hi,
I'm trying to do load balancing between 2 asterisk servers using SIP
load balancer,
I beleive what you're looking for is a scalable SIP proxy, like SER :)
That way, all clients registers to SER and SER redirects the caller to
one of the asterisk boxes. Search the wiki at voip-info.org for
asterisk at large :)
Yes, that is one of the many pages I've read. But we still have a
Hi!
http://drmac.homeunix.net/images/load_balancer.jpg
You won't need the second balancer. SER can do that.
Seconded.
For growth, all you do is add more SER and more Asterisk boxes.
Are you sure one SER box won't be sufficient?
Makes sense to me to have these TWO - you can take one
For growth, all you do is add more SER and more Asterisk boxes.
Are you sure one SER box won't be sufficient?
Makes sense to me to have these TWO - you can take one of those
off-line
without interrupting service, and that's the entire idea of this
discussion, isn't it? ;-
Yeah
Get two cisco load
Inline...
I've read several other emails and pages on the wiki but none give any
deffinate answers. if you have 20 asterisk servers each with 4 pri's, all
running RealTime Extensions and RealTime SIPBuddies from the same MySQL
server, what prevents you from putting all 20 servers behind a
I'd have to guess that registrations would be the tricky part of an
implementation simply because there are so many variations of that.
Actually, this is the easiest part. It doesn't matter how often a UA
registers nor does it matter to which of the 20 servers handles the
registration since
I'd have to guess that registrations would be the tricky part of an
implementation simply because there are so many variations of that.
Actually, this is the easiest part. It doesn't matter how often a UA
registers nor does it matter to which of the 20 servers handles the
registration
I've read several other emails and pages on the wiki but none give any
deffinate answers. if you have 20 asterisk servers each with 4 pri's,
all
running RealTime Extensions and RealTime SIPBuddies from the same MySQL
server, what prevents you from putting all 20 servers behind a single
load
I beleive what you're looking for is a scalable SIP proxy, like SER :)
That way, all clients registers to SER and SER redirects the caller to
one of the asterisk boxes. Search the wiki at voip-info.org for
asterisk at large :)
Yes, that is one of the many pages I've read. But we still
Non-Commercial
Discussion" asterisk-users@lists.digium.com
Sent: Wednesday, February 02, 2005 5:43
PM
Subject: Re: [Asterisk-Users] load balancing 20
asterisk servers
I beleive what you're looking for is a scalable SIP proxy, like
SER :) That way, all clients registers to SER and SER redirects
Hi,
You may want to look into LVS (Linux Virtual Server). It allows load
ballancing in a highly configurable way.
http://www.linuxvirtualserver.org/
We use it on our web and mail server to load ballance across multiple
hosts. The way we have it configured
it will maintain a session for 15
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