[OpenSIPS-Users] Question about using load balancing (dispatcher) within failure_route

[Klaus Darilion]

IIRC I also had problems some time ago in combination with 
force_send_socket (I think append_branch() did not copied the socket info)

klaus

Bogdan-Andrei Iancu schrieb:
> Hi Woody,
> 
> I will fix this bug shortly and then you can easily use DISPATCHER with 
> round-robin, according to your needs.
> 
> Thanks for report,
> Bogdan
> 
> Woody Dickson wrote:
>> Hi Bogdan,
>>
>> Thank you for your prompt reply. 
>>
>> Rightnow, I can hard code ONE B2BUA in the failure_route, but I would 
>> rather have multiple B2BUA dispatched in a round robin fashion when 
>> the failure_route is hit and have the IPs extracted from DB.
>>
>> Is there any possible workaround that can resolve this problem for the 
>> short term?
>>
>>
>> Thanks alot.
>>
>> Woody
>>
>> On Mon, Oct 20, 2008 at 3:46 PM, Bogdan-Andrei Iancu 
>> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>>
>>     Hi Woody,
>>
>>     This is related to a bug I recently found in the dispatcher module
>>     - the ds_next_dst() function, even if it should set only the
>>     destination URI, it internally creates a branch - so, you will end
>>     up with 2 branches actually :
>>       1) first, with the original URI and the DST from dispatcher
>>       2) second, with the URI from script (rewrite) and no dst;
>>
>>     Regarding the failover support, you need, to set some params, like:
>>       dst_avp -
>>      http://www.opensips.org/html/docs/modules/devel/dispatcher.html#id2529512
>>       grp_avp -
>>     http://www.opensips.org/html/docs/modules/devel/dispatcher.html#id2533853
>>       cnt_avp -
>>     http://www.opensips.org/html/docs/modules/devel/dispatcher.html#id2509422
>>
>>     Regards,
>>     Bogdan
>>
>>     Woody Dickson wrote:
>>
>>         Hello,
>>
>>         I have my dispatcher setup fine for my INVITE so Opensips can
>>         now distribute calls to one of the two B2BUA.  What I would
>>         also like to do is that when 486 is recevied, Opensips can
>>         branch another route back to one of the two B2BUA which will
>>         play the voicemail.   I tried:
>>         ds_select_dst("1", "4");
>>         ds_next_dst();
>>         rewritehostport("192.168.1.101:5070
>>         <http://192.168.1.101:5070> <http://192.168.1.101:5070>");
>>
>>
>>         only rewritehostport works, but I don't want to hard code the
>>         host and port inside the config script.  If I use
>>         ds_next_dst(), I am getting "WARNING:dispatcher:ds_next_dst:
>>         failover support disabled".  Does anyone know how to resolve
>>         this problem?
>>
>>         Here is my failure_route block:
>>
>>         failure_route[1] {
>>
>>               if (t_was_cancelled()) {
>>                    exit;
>>            };
>>            if (t_check_status("486")) {
>>                   revert_uri();
>>                      prefix("b");
>>                      xlog("L_ERR","Stepped into the 486 ruri=<$ru>");
>>                     #ds_select_dst("1", "4");
>>                    ds_next_dst();
>>                #rewritehostport("192.168.1.131:5070
>>         <http://192.168.1.131:5070> <http://192.168.1.131:5070>");
>>
>>                     append_branch();
>>                     route(1);
>>                            exit;
>>            };
>>
>>
>>         Thanks in advance for all your help.
>>
>>         Regards,
>>         Woody
>>
>>         ------------------------------------------------------------------------
>>
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>>
>>
>>
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