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 To :  "'AmberVoIP'" <ambervoip@g...>,"'Daniel Salama'" <lists@i...>
 From :  "Bill Kunyiha" <bill@f...>
 Subject :  RE: [yate] Billing and others-IP Based Authentication
 Date :  Thu, 3 Aug 2006 18:18:36 -0700
Anything that comes in as Sip or H323 comes through ${module}h323=goto
from_h323 and ${module}sip=goto from_sip which sends it to from_h323 or
from_sip where we do ip based authentication.
Anything from ip 192.168.1.3 is sent to sip_trunk1 while anything from
192.168.1.5 is sent to sip_trunk2

${module}h323=goto from_h323
${module}sip=goto from_sip


[from_sip]
${address}^192\.168\.1\.3=goto sip_trunk1
${address}^192\.168\.1\.5=goto sip_trunk2

[sip_trunk1]
^1\(.*\)$=sip/sip:@questcommunications 

[sip_trunk2]
^\(.\)\(.\)\(.\)\(.\)$=sip/sip:@pacwestcommunications


[from_asterisk_h323]
^1\(.*\)$=sip/sip:@6... ; callername = bill.kunyiha

Bill 

-----Original Message-----
From: AmberVoIP [mailto:ambervoip@g...] 
Sent: Thursday, August 03, 2006 2:58 PM
To: Daniel Salama
Cc: yate@v...
Subject: Re: [yate] Billing and others

On Thu, 2006-08-03 at 16:40 -0400, Daniel Salama wrote:
> Thanks for the prompt response.
> 
> I take your word that these billing packages support Yate, even  
> though none of them announce it on their site. So far, the most  
> complete one I saw was DTL. However, that seems to be a commercial  
> solution and I'm waiting to get a quote from them.

yate support radius and attributes, in theory, you can use any
radius-billing with it.
> 
> As far as the call center solution is concerned, I don't mind using  
> Asterisk. I currently use it for that. However, I was looking for a  
> more robust platform for wholesale traffic softswitching that can do  
> SIP and H323 and that's why I'm considering Yate, since I've had a  
> few issues with Asterisk on large volume environment, especially  
> handling H323 traffic.

same. i am not from yate team and still studing product. i come to yate,
since have problems with asteirsk & h323. i still have issues, for
example, h323 -> yate -> h323 do not work with Hypermedia (voip
gateway), but yate is much bettter than asterisk h323 realisation. it
good for most cases.
> 
> One thing I have not been able to figure out in Yate, is how to do IP- 
> based authentication and routing. For example, for wholesale  
> customers that we authorize based on IP-address, we need to be able  
> to restrict their traffic in simple ways, such as where are they  
> allowed to terminate. Any help would be greatly appreciated.

same as asterisk, yate have 'account file', where you can put all
accounts, can do also ip-based restrictions.
in harders situation, you can use call handler, which is like asterisk
AGI, so you can keep config in database.... hope by tomorrow , you will
receive detailed response, but all that is possible.

Andy.
> 
> Thanks,
> Daniel
> 
> On Aug 3, 2006, at 3:26 PM, AmberVoIP wrote:
> 
> > On Thu, 2006-08-03 at 13:52 -0400, Daniel Salama wrote:
> >> Hi all,
> >>
> >> Is there any billing platform for Yate that supports both SIP and
> >> H323 as well as interfacing with Radius? Ideally, it should handle
> >> multi-server configurations, maybe all servers pointing to a single
> >> MySQL server. Also, open source would be preferable for we may need
> >> to modify/enhance/customize.
> >>
> > http://ibs.sourceforge.net
> > http://www.ibsng.com
> > also i tested porta and mind with yate
> > from hosted - friend of mine tested with DTL (www.datatechlabs.com)
> >
> >> Also, can Yate be used in call center applications? To be more
> >> specifically, can it support agent login/logout/suspend, queueing,
> >> campaign-based music on hold, etc?
> >>
> >
> > Yes, but you must do lot of development to this area. Now i using yate
> > for wholesale traffic, transit, some call progress, so for me yate is
> > softswitch and gatekeeper.
> >
> > For call center and voice processing i like asterisk pbx.
> >
> > Andy.
> >
> >
> >
>