Hi Stan, Have you seen - http://lists.opensips.org/pipermail/users/2010-April/012414.html - this it about attended transfer when doing balancing for Asterisk.
Regards, Bogdan Stanisław Pitucha wrote: > On 05.05.2010 11:50, Paris Stamatopoulos wrote: > >> Has anyone experience this before? Do you have any working solutions? >> > > Yes and yes. Short version: It's impossible without hacking asterisk. > > Long version: > First we tried routing all related calls to the same destination using > the dialog module. It would work, but only in the most trivial cases. It > will work only if you can always match the destination or the source > number to an existing call. If you use call forking / huntgroups / > queues / redirection on your asterisk - forget about easy transfers - > you will get calls which don't match on any field, but are still going > to the same phone. > > Now - theoretically transfers should work between asterisks based on the > information in refer-to - it already includes the host which handles the > call and in the transfer context you can extract that address and force > a Dial() to that destination. I never discovered why it doesn't quite > work. Half of the calls failed and sometimes I got deadlocks (still > using 1.4 - maybe 1.6 is better here). > AFAIK there's also no real possibility to set the callerid in that case > to figure out who initiated the transfer. > > But we've got a working solution now: > - The first part is a custom res_cluster module which will publish for > each call (hooked into the Dial() command) the callid, ftag, ttag and > the ip of the local asterisk into a shared mysql database. > - The second part is activated on refer. It does some magic in > handle_refer() (or something like that) and then magicks the call to > magically know the correct callerid, customer id, checks whether the > transfer is allowed and dials the destination asterisk with correct > refer-to headers. (destination comes from the database) > > It could be much simpler if you don't have to handle different customers > and bill them separately. > > Maybe I missed some other option... but that's the solution we got to > after ~3 months of testing different methods, annoying asterisk mailing > list and their IRC. And it works just fine so far. > > I'm not sure I can release all the needed patches, but if you want them, > let me know. > > Stan > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Bogdan-Andrei Iancu www.voice-system.ro _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
