[Asterisk-Users] Need some help

[Alex]

The reason i am using 1 scenario is because of routing, authentication and accounting. If i can use these things in asterisk i will use it.
 
1)What is the best way to do that, through extensions ?
 
2)When the call coming i can check the phone number and if the number should go to pstn i will forward this to asterisk. now i need example how i can forward the call to pstn in extension.cfg (or there are other way to forward the call to pstn inside asterisk).
 
3) Another question if all my users authenticated with ser how i can send call to the other user who included in the same database and connected to the same SER server (LOCAL CALLS) should not go to the pstn. 
Should i use softphone(user1) ->SER->ASTERISK->SER->softphone(user2)
 or i need to use softphone(user1)->SER-(user2) if i use this scenario i will loose the cdrs of the asterisk.
 
The reason i am asking the these question because till now i didn't use asterisk and i forwarded the call through ser and it's working fine. I wanted to use IVR system so i installed the asterisk and also asterisk has the CDRs. now i need to use this scenario 
long distance call:   softphone -> SER -> Asterisk -> pstn (long distance calls)
local calls: softphone->SER ->Asterisk -> SER->softphone ( I am not sure if i can do that without registering users inside sip.cfg in the asterisk.)
 
Any help will be appreciated.
 

Yair Hakak <yhakak at gmail.com> wrote:
Duh, i'm an idiot. I meant scenario #1.

-yair


On Wed, 23 Mar 2005 18:52:28 +0200, Yair Hakak wrote:
> Hello,
> what is the benefit of your scenario #2? I'm not understanding what
> it adds for you...
> 
> -yair
> 
> 
> On Wed, 23 Mar 2005 08:49:37 -0800 (PST), Alex wrote:
> > Hi all
> >
> > I have a couple of questions maybe you guys can help me with them
> >
> > I have sip phones , SER server , Asterisk.
> >
> > what is the best way to do that (also with accounting and authentication).
> >
> > which one of those options
> > 1) sipphone -> SER -> ASTERISK -> SER -> PSTN
> >
> > 2) sipphone -> SER ->ASTERISK ->PSTN
> >
> > on the first option i am trying to return the call to the ser after it's
> > pass the asterisk for some routing solutions and accounting. but i have some
> > problems to hear the other side.
> >
> >
> > Thanks for any advice
> >
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