Hi Stan Again. Thanks very much for the quick response.
I've setup 2 Domains, Domain A and Domain B with usera in Domain A and userb in Domain B. I'm experiencing a strange problem. When I do "opensipsctl ul show userb" it shows that the Location for this user is in Domain A. FIFO command was: :ul_show_contact:opensips_receiver_7516 location us...@domaina.com When I do "opensipsctl ul show us...@domainb.com" it shows that the Location for this user is in Domain B. FIFO command was: :ul_show_contact:opensips_receiver_9434 location us...@domainb.com I'm able to setup a call between the 2 users, but the call drops after +-30 seconds. VOICE data is sent and received between the devices while in the +-30 seconds call duration. Any suggestions? Should i send you my config script? Thanks again for the help. Regards Deon On 30 Sep 2010, at 1:19 PM, Stanisław Pitucha wrote: > On 30/09/10 11:59, Deon Vermeulen wrote: >> Let me try and explain a scenario as brief as I possibly can. >> >> I have Company A (Domain A) and Company B (Domain B). >> >> Domain A and B should be completely Transparent to each other. >> >> See like Domain A and B in their own respective Bubbles completely >> separated from each other, each with functions like Hunt groups, >> IVRs, >> Conferencing, Presence, Music on Hold, etc... > That's fine with domains. Just make sure usrloc / aliases use domains. > (there's a mod_param for that) and that the domain names are filled > correctly in the db. > >> Extension 123 should be able to exist in both Domains and make/ >> receive >> calls. > Handled by domains - usrloc / registrar should handle it just fine. > Run > standard lookup("location") before relaying the packet and it will go > the the correct user if he's registered. Make sure that the users in > table `subscribers` use a different domain. > >> Domain A and B should not be able to make any VOICE calls between >> each >> other, but via their respective PSTN/GSM Gateways, unless they are >> put >> in a Group/Class allowing them. Both Bubbles in a bigger bubble. > Aliases will help you resolve that. Bind PSTN numbers to > extensi...@domain and lookup("aliases") before sending the call to > PSTN > numbers in order to loop back to the correct users. So basically > something like: > if (lookup("aliases")) { route(local_call); } > else { route(external_call); } > > route[local_call] { > if(lookup("location")) { t_relay(); } > else { return some error, probably 404 } > } > >> Each Domain should have their own respective PSTN, GSM, etc Gateways >> and be completely transparent to the other Domain, ie. Domain A >> should >> not be able to use any of Domain Bs gateways to make/receive calls >> and >> vise-versa. > I think this can be handled by drouting - I'm not using it, so not > 100% > sure. If it doesn't, then you can map domain names onto dispatcher > groups and use ds_select_domain() instead. > > As for the received calls, I'm not sure if you want to limit anything? > If you're only going to get pstn numbers, do you really care if you > got > a call from the "wrong" gateway? If you do, just check the source > against the domain after lookup("aliases"). If you don't - your > incoming > call looks just like your outgoing one. (apart from the > route(external_call) of course - that should be 403 or something > like that). > > Regards, > Stan > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users