[Asterisk-Dev] Call authentication in Asterisk

[Aram Ter-Martirosyan]

	I have seen a lot of people talking about using asterisk as a prepaid
calling card platform.  I was wondering how would one perform call
authentication in Asterisk -
1.Is there a Radius support in Asterisk?
2.What other authentication mechanism can be used with Asterisk?
3.Is it possible to use Asterisk as IVR system for Prepaid calling platform?
Also I would like to find out if it is possible to use Asterisk as H323
and/or SIP gatekeeper/Proxy server to not actually have the media stream
flow through Asterisk but just connect the endpoints together?

	Thank you in advance for your help

Aram Ter-Martirosyan
Senior Account Manager
Hi-Tech Gateway, Inc.
http://www.hi-teck.com
1225 Grand Central Ave.
Glendale, CA 91201
aram at hi-teck.com
tel 818.546.4601
fax 818.546.4617
Turning Technology Into Business Solutions


-----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com]On Behalf Of
timecop at pbx.mine.nu
Sent: Sunday, July 27, 2003 2:03 AM
To: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] SIP Session-Timer support in asterisk


weird Ive been trying to send this mail for hte last few weeks and it
isnt getting through... another repost..

According to this document,

http://www.ietf.org/internet-drafts/draft-ietf-sip-session-timer-11.txt

This is some rather new feature designed to timeout idle SIP sessions.
Unfortunately, the SIP provider I use decided to implement this "feature".
Asterisk does not support it.
I (temporarily) hacked it by adding required headers into transmit_invite,
however that cancels the call after Session-Expires number of seconds,
because INVITE/UPDATE isnt sent back in that period.

Anyone has same problem and knows enough about asterisk internals to
develop support for this "feature"?

The real reason I think these guys are implementing support for it is
because here SIP is billed per minute for all calls, so they would
naturally want to timeout sessions isntead of billing for hours of calls
that were over long time ago...

thanks
tim
_______________________________________________
Asterisk-Dev mailing list
Asterisk-Dev at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-dev