[OpenSIPS-Users] opensips+dispatcher+asterisk problem

Bogdan-Andrei Iancu bogdan at voice-system.ro
Mon Jul 13 16:24:01 CEST 2009


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
>  
>  




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