[OpenSIPS-Users] a little help

[Bogdan-Andrei Iancu]

Hi Albert

Albert Paijmans wrote:
> Thanks,
>
> My background is old analog telephony systems and I know a little bit 
> of Asterisk.
> With Asterisk we do not allow reinvites and it does most things we 
> want. And it is relatively easy to secure an Asterisk server. 
I wouldn't bet on it
> Reinvites are to make sure nat issues do not show up.
neither on this.
>
> What I am trying to do is set up a proxy and registrar for 100.000 
> subscribers. It's behaviour should be stateful. It should allow 
> anonymous invites for subscribers (extensions) and alias database. So 
> you could call form Ekiga.net to our domain. Our subscribers 
> (extensions) should be authenticated and allowed to dial other domains 
> and pstn. Because of nat issues every call must be send to Asterisk.
>
> So after an invite from subscriber 101 to subscriber 102 (usrloc) 
> dialplan rewrites to usrloc, usrloc is then translated to loadbalancer 
> and send out to an Asterisk server.
>
> No matter where you are dialing to you will always be forwarded to a 
> media server for rtp.
> That's why I asked about the t_onreply function.
but what is the actual problem you are facing (that makes you think that 
onreply() is the answer) ? - the onreply route is used only for 
inspecting the received replies, without any routing power.
>
> I am sorry that I am not so good in this programming language. :(
We all learn each day :)

Regards,
Bogdan
>
>
> Albert
>
>
> On Wed, Jun 2, 2010 at 8:48 PM, Bogdan-Andrei Iancu 
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>
>     Hi Albert,
>
>     I do not fully understand what you want to achieve - maybe
>     describing a
>     simple flow (logical point of view) will really help in helping you.
>
>     Best regards,
>     Bogdan
>
>     Albert Paijmans wrote:
>     > Hi,
>     >
>     > I am a bit at a loss right now, I have a script wich does dialplan
>     > translations but permission and group modules are loaded,
>     registration
>     > fails.
>     > The dialplan module and avpops should give an attribute to a number.
>     > Also I am wondering (since the script does not work I can''t test
>     > this) if a t_relpy route on usrloc would work.
>     > So after OpenSIPS does database lookup for extensions and db_alias
>     > sends an invite and relay both extensions to an Asterisk server via
>     > gateway list.
>     >
>     > I have send the opensips.cfg as attachement
>     >
>     > Thanks
>     >
>     > Albert
>     >
>


-- 
Bogdan-Andrei Iancu
www.voice-system.ro