[asterisk-users] load balancing SIP extensions
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 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 not registered here then try to look it up on *2 and establish the call there 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. So in my case where both *1 and *2 have the same local extension range 4XXX, can I go the DUNDi route or should I stop bashing my head on that and explore another solution? If someone has configured a similar system then I'd greatly appreciate some tips. I read a few dundi docs like http://www.voip-info.org/wiki-DUNDi. Thanks Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
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 not registered here then try to look it up > on *2 and establish the call there You can do that using the dial plan. - Create an IAX link between both servers - DIal plan in both servers: First priority Dial using SIP/EXTEN Second priority IAX/EXTEN Dial IAX/EXTEN -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
> 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 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 not registered here then try to look it up > on *2 and establish the call there >... I've tried doing something similar and came with two options. The common to them is that I use MySQL for realtime extensions, and set "systemname" parameter to the IP address of the server where the phone registers. 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 process the call and the audio. - Send a "refer" message to the caller to contact the other server. I had this working in the lab but not in production yet. If you want the dialplan code for this then email me. __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
--- 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 establish a call if > it's > > REGISTERED here > > - if it's not registered here then try to look it > up > > on *2 and establish the call there > > You can do that using the dial plan. > > - Create an IAX link between both servers > - DIal plan in both servers: > First priority Dial using SIP/EXTEN > Second priority IAX/EXTEN Dial IAX/EXTEN Thanks. I'll try that although I hope it won't go into an infinite loop between the 2 servers. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
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 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 not registered here then try to look it > > up > > > on *2 and establish the call there > > > > You can do that using the dial plan. > > > > - Create an IAX link between both servers > > - DIal plan in both servers: > > First priority Dial using SIP/EXTEN > > Second priority IAX/EXTEN Dial IAX/EXTEN > > Thanks. I'll try that although I hope it won't go into > an infinite loop between the 2 servers. > > > > > > > Looking for last minute shopping deals? > Find them fast with Yahoo! Search. > http://tools.search.yahoo.com/newsearch/category.php?category=shopping > > ___ > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
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 comes from IAX it means that PHONE is not registered in the other server. Just create special extensions to take the IAX calls (instead of GoTo): PHONE is 101 SERVER 1 exten => 101,1, Dial SIP/101 exten => 101,1, Dial IAX-SERVER2/55101 exten => 55101,1, Dial SIP/101 exten => 55101,1, Hangup SERVER 2 exten => 101,1, Dial SIP/101 exten => 101,1, Dial IAX-SERVER1/55101 exten => 55101,1, Dial SIP/101 exten => 55101,1, Hangup I hope it helps, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
> > 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 what you describe. http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
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 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 not registered here then try to look it up > on *2 and establish the call there > > 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. > So in my case where both *1 and *2 have the same local > extension range 4XXX, can I go the DUNDi route or > should I stop bashing my head on that and explore > another solution? > > If someone has configured a similar system then I'd > greatly appreciate some tips. > I read a few dundi docs like > http://www.voip-info.org/wiki-DUNDi. > > Thanks > > > > > > Looking for last minute shopping deals? > Find them fast with Yahoo! Search. > http://tools.search.yahoo.com/newsearch/category.php?category=shopping > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Have you tried placing the sip registrations in a db using realtime? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
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 > process > the call and the audio. > > - Send a "refer" message to the caller to contact the other server. > You may actually want to use a "redirect" message for this (e.g SIP 302 response). In any case, traversing only one server in the signaling/media path as opposed to two would generally seem more efficient. -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
--- 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 not). I'll take a look at the pdf link of the previous post. Thanks Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing SIP extensions.
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, statically defined in the config file of the
phone, etc.).
The idea is to locate on which server the destination phone is registered and
redirect the call to it. For this:
/etc/asterisk/asterisk.conf has the parameter "sysname" set to its IP address
(you can use also a DNS name, but I want to be independent of name
resolution). This causes the server to set the field "regserver" to be
saved in the MySQL database to the IP address of the server.
/etc/asterisk/extensions: The logic to check whether the value of "regsever"
is different from "sysname" and if so - redirect the call. The code
fragments are (I am using AEL):
To get the regserver from the database:
MYSQL(Query resID ${connid} SELECT regserver from sip_users where
name='${EXTEN}');
MYSQL(Fetch FetchId ${resID} RegServer);
MYSQL(Clear ${resID});
so now RegServer contains the server where the phone is registered. Next:
if(("${DEVSTATE(SIP/${EXTEN})}" == "UNAVAILABLE") ||
("${DEVSTATE(SIP/${EXTEN})}" == "INVALID")) {
if("${SYSTEMNAME}" != "${RegServer}") {
Transfer(SIP/[EMAIL PROTECTED]);
return;
};
};
I check for the device state so in case the phone has double registration
(primary and backup server) it will be processed localy.
Regards, __Yehavi:
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