Re: [OpenSIPS-Users] Opensips + Dispatcher + asterisk + fail over problem

2009-08-06 Thread urmi lakkad 
Hello,

Yes, That was my mistake. now my asterisk fail over mechanism is working
fine.

Thanks a lot for helping me.

-urmi

On Tue, Aug 4, 2009 at 5:28 PM, Saúl Ibarra  wrote:

> It's OPTIONS, not OPTION :)
>
>
> --
> Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de
> disketes."
> 
> http://www.saghul.net/
>
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Re: [OpenSIPS-Users] Opensips + Dispatcher + asterisk + fail over problem

2009-08-04 Thread Saúl Ibarra 
It's OPTIONS, not OPTION :)


-- 
Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes."

http://www.saghul.net/

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Re: [OpenSIPS-Users] Opensips + Dispatcher + asterisk + fail over problem

2009-08-04 Thread urmi lakkad 
Hello Alex,

First of all Thanks for ur Attention and quick response.

 If I use the OPTION then on Asterisk I m getting following:

[Aug  4 17:15:32] NOTICE[7765]: chan_sip.c:14958 handle_request: Unknown SIP
command 'OPTION' from '192.168.1.30'

Even when that Asterisk Comes up, opensips is not sending calls to that
Asterisk.


-Urmi


On Tue, Aug 4, 2009 at 5:07 PM, Alex Balashov wrote:

> urmi lakkad wrote:
>
>  modparam("dispatcher", "ds_ping_method", "INFO")
>>
>
> Asterisk does not respond to these.  Try using the OPTIONS method instead.
>
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (678) 237-1775
>
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Re: [OpenSIPS-Users] Opensips + Dispatcher + asterisk + fail over problem

2009-08-04 Thread Alex Balashov 
urmi lakkad wrote:

> modparam("dispatcher", "ds_ping_method", "INFO")

Asterisk does not respond to these.  Try using the OPTIONS method instead.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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[OpenSIPS-Users] Opensips + Dispatcher + asterisk + fail over problem

2009-08-04 Thread urmi lakkad 
Hello,

I am using Opensips and Asterisk for my call flow.

I am using 3 Asterisks for call forwarding. Opensips's dispatcher module is
doing the task of load balancing among all 3 Asterisk servers in a round
robin fashion. i. e.

1st call to 1st Asterisk
2nd call to 2nd Asterisk
3rd call to 3rd Asterisk

If any 1 of the Asterisk goes down,(i.e fail to respond), then dispatcher
stops sending calls to that particular Asterisk. Its working fine.

But again when that Asterisk comes up( i.e comes to network or become live)
, the dispatcher should start sending calls to that Asterisk server. Can u
please suggest me how can I achieve this ?

Following is my opensips.cfg file


loadmodule "dispatcher.so"

modparam("dispatcher", "list_file",
"/usr/local/etc/opensips/dispatcher.list")

modparam("dispatcher", "flags", 2)
modparam("dispatcher", "dst_avp", "$avp(i:271)")
modparam("dispatcher", "grp_avp", "$avp(i:272)")
modparam("dispatcher", "cnt_avp", "$avp(i:273)")

modparam("dispatcher", "ds_ping_method", "INFO")
modparam("dispatcher", "ds_ping_interval", 1)
modparam("dispatcher", "ds_probing_mode", 1)

route{

if (is_method("INVITE"))
{
  if (nat_uac_test("16"))
  {
  fix_nated_contact();
  force_rport();
};


ds_select_dst("1", "0");
t_on_reply("1");
t_on_failure("1");
forward();
exit;

}

 
Dispatcher.cfg

1 sip:192.168.1.1:5060 #Asterisk-1
1 sip:192.168.1.1:5061 #Asterisk-2
1 sip:192.168.1.1:5062 #Asterisk-3



 Thanks for your attention.

-Urmi
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