[OpenSIPS-Users] Paid Consultation Request

[Geoffrey Mina]

Thanks for the input.  I did test the 408 scenario by disabling one of
the asterisk systems.  I was connected properly and the one end point
was removed from the pool.  I will run some more tests and analyze the
debug info more carefully to see exactly what path is being taken.

In regards to asterisk as a PSTN gateway, it isn't acting as such, so
I don't have to worry about that.  The systems are simply providing
end point IVR functionality.  Asterisk will ALWAYS answer the call
itself.

I will definitely add the 500 and 600 series errors to my failover handling.

Excellent Input!!!

Much appreciated.

On Wed, Feb 11, 2009 at 5:10 PM, Iñaki Baz Castillo <ibc at aliax.net> wrote:
> I don't understand the following:
>
>
> -----------------------------------------------
> ########################################################################
> ## Handles relay of INVITE messages
> ## with round-robin load balancing
> ########################################################################
> route[1]{
>        ds_select_domain("1","4");
>        t_on_reply("1");
>        t_on_failure("1");
>        t_relay();
> }
>
> [...]
>
> #######################################################################
> ## Handles failure of INVITE forwarding
> #######################################################################
> failure_route[1]{
>        xlog("L_INFO","Failure route, trying again\n");
>
>        if(t_check_status("408")){
>                xlog("L_INFO","Got a 408 Timeout, flagging dest as invalid\n");
>                ds_mark_dst();
>                route(1);
> ----------------------------------------------
>
>
> In failure_route[1] you call route[1], and in route[1] you
> execute "ds_select_domain()".
> According to the doc:
>  http://www.opensips.org/html/docs/modules/devel/dispatcher.html#id271282
> "ds_select_domain()" can only be called from REQUEST_ROUTE, and not from
> FAILURE_ROUTE.
> I wonder why you don't get an error when starting your OpenSIPS. ¿?
>
> I expect you should use "ds_next_domain" in FAILURE_ROUTE:
>  http://www.opensips.org/html/docs/modules/devel/dispatcher.html#id271328
>
> Are you 100% sure that this script works in your lab? How have you simulated
> the "408" from a server?
>
>
> Other point: you consider the case of a 408 to dissable a gateway. I've worked
> with OpenSIPS in front of an Asterisk acting as PSTN gateway using PRI.
> Sometimes, when the PSTN called doesn't answer after XX seconds, the telco
> replies a ISUP code that Asterisk converts to "408". Be careful because if
> that occurs your OpenSIPS will mark that Asterisk as "dead".
>
>
> Other point: why do you only consider 408 to dissable a gateway? Imagine that
> the Asterisk has a problem and replies 500 (Internal Error) or 503. But if
> you add those casees, be sure of adding "Hangup" at the end of each possible
> extension in your Asterisk dialplan. If not, Asterisk will reply 503.
>
>
>
>
>
>
>
> --
> Iñaki Baz Castillo
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>