[OpenSIPS-Users] opensips+dispatcher+asterisk problem

[Bogdan-Andrei Iancu]

Hi Ram,

By default, if failover is enabled, Dispatcher module will try all 
destinations from the set, until it finds one working. You need to catch 
the failures in failure_route and to use ds_next_domain|dst() functions 
to try the next available destinations.

Also, by using the pringing option, you can re-enable automatically 
destinations that came back online.

Regards,
Bogdan

ram wrote:
>
>
> On Tue, Jun 30, 2009 at 5:20 PM, Bogdan-Andrei Iancu 
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>
>     Hi Ram,
>
>     I found your email on the Asterisk mailing list also ;)
>
>     So, to answer here also: do you get any reply back from Asterisk ?
>
>  
> Hi Bogdan
>  
> thanks for the reply
>  
> I have made a quick Fix, iam not sure how far its good.
>  
> Just put coment in  secret , in the Asterisk 
> Additional_a2billing_sip.conf. rather doing twise  authentication.
>  
>  
> But i have another problem here with the Dispatcher,
> dispatcher sending calls round robin,
>  
> 1 st call to 1st *
> 2nd call to 2nd *
> 3 call to 3rd *
>  
> if 2nd Asterisk fails to respond still Dispatcher module sending calls 
> to 2nd asterisk
>  
> how can i fix this issue with Dispatcher, if any one of * box not 
> reachable it should detect and send call to 3rd *
>  
> if 2nd comes back in to network and live, it should send to 2nd *
>  
> how can i achive this ?
>  
> Ram
>  
>