[OpenSIPS-Users] opensips+asterisk call dropping in 20 seconds

[Bogdan-Andrei Iancu]

Hi Ram,

a call drop (at signalling level) after 20 secs  typically shows a 
problem with the ACK - the ACK does not get back to callee and callee 
bye's the call as it never think the call was not confirm.

Could you post a trace for the KO call ?

Regards,
Bogdan

ram wrote:
> Hi
>  
> In continuation with the subject
>  
> when i call intiated from Opensips the call drop in 20seconds
>  
> but when i register directly from * box i dont see the call drop even 
> for 20-30min of talk
>  
> any suggestions
>  
> Ram
>
> On Tue, Jun 30, 2009 at 8:35 PM, ram <talk2ram at gmail.com 
> <mailto:talk2ram at gmail.com>> 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
>      
>      
>
>