Hi Ronald,<br><br><div><span class="gmail_quote">On 9/19/06, <b class="gmail_sendername">Ronald Wiplinger</b> <<a href="mailto:ronald@elmit.com">ronald@elmit.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I am thinking if re-invite will interfere accounting.</blockquote><div><br>No it won't <br></div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Please help me to figure it out:<br><br>Phone A is registered at asterisk and calls a gateway. If the gateway<br>allows re-invite than the rtp would go directly from phone A to the<br>gateway, while the sip messages are still going through Asterisk.
<br>Asterisk will be informed when the call ended.<br>If it is a postpaid accounting, just bill the customer, however, how is<br>it for a pre-paid (calling card user)?<br>I think Asterisk will have no power to turn off the call from A to the
<br>gateway.<br>Even more, if the gateway would allow to end a call and continue with a<br>new call, the new call would not be billed (or would it)?</blockquote><div><br>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.
<br></div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I guess the solution must be re-invite=no<br>However, re-invite=no means that each call is going with rtp also
<br>through my server, what means for a remote phone, I have to provide for<br>both legs the bandwidth.</blockquote><div><br>Personally, I would always handle the RTP on quality/accountability/consitency grounds but, yes, that will incur a bandwidth overhead.
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