Re: [asterisk-users] Accounting and re-invite

2006-09-19 Thread Simon Woodhead
Hi Ronald,On 9/19/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I am thinking if re-invite will interfere accounting.No it won't 
Please help me to figure it out:Phone A is registered at asterisk and calls a gateway. If the gatewayallows re-invite than the rtp would go directly from phone A to thegateway, while the sip messages are still going through Asterisk.
Asterisk will be informed when the call ended.If it is a postpaid accounting, just bill the customer, however, how isit for a pre-paid (calling card user)?I think Asterisk will have no power to turn off the call from A to the
gateway.Even more, if the gateway would allow to end a call and continue with anew call, the new call would not be billed (or would it)?The SIP messages control the call, irrespective of where the RTP goes so Asterisk can terminate/set-up calls exactly as if the RTP was being handled. 
I guess the solution must be re-invite=noHowever, re-invite=no means that each call is going with rtp also
through my server, what means for a remote phone, I have to provide forboth legs the bandwidth.Personally, I would always handle the RTP on quality/accountability/consitency grounds but, yes, that will incur a bandwidth overhead.

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[asterisk-users] Accounting and re-invite

2006-09-18 Thread Ronald Wiplinger

I am thinking if re-invite will interfere accounting.

Please help me to figure it out:

Phone A is registered at asterisk and calls a gateway. If the gateway 
allows re-invite than the rtp would go directly from phone A to the 
gateway, while the sip messages are still going through Asterisk. 
Asterisk will be informed when the call ended.
If it is a postpaid accounting, just bill the customer, however, how is 
it for a pre-paid (calling card user)?
I think Asterisk will have no power to turn off the call from A to the 
gateway.
Even more, if the gateway would allow to end a call and continue with a 
new call, the new call would not be billed (or would it)?


I guess the solution must be re-invite=no 
However, re-invite=no means that each call is going with rtp also 
through my server, what means for a remote phone, I have to provide for 
both legs the bandwidth.


Would here a rtpproxy or mediaproxy  help? If how and why?

bye

Ronald


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