i thought so! this helps alot. thanks so much! <BR><B><I>Yair Hakak <yair@hakak.com></I></B> wrote: <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"> <DIV>the thing to remember is that these terms are from the point of view of the PSTN. </DIV> <DIV>So, SIP origination is Direct Inbound Dial (DID, or DDI in european parlance) which allows callers on the PSTN to originate calls that end up at your SIP server.</DIV> <DIV>SIP termination allows calls which originate on your SIP server to terminate on the PSTN, i.e. to reach a non-voip line.</DIV> <DIV> </DIV> <DIV>Voip providers who provide "plans" are bundling 2 distinct services. Broadvoice, for example, does not expect its users to understand the terms, and just offers them what used to be called a "phone line" - the ability to make calls (termination) and recieve them (origination). </DIV> <DIV> </DIV> <DIV>i hope this helps,</DIV>
<DIV> yair<BR><BR> </DIV> <DIV><SPAN class=gmail_quote>On 9/10/06, <B class=gmail_sendername>Christopher Corn</B> <<A href="mailto:christopher_corn@yahoo.com">christopher_corn@yahoo.com</A>> wrote:</SPAN> <BLOCKQUOTE class=gmail_quote style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"> <DIV> <DIV>can someone please explain the differnces to me??? </DIV> <DIV> </DIV> <DIV>I have an asterisk system im setting up for a small office (4 or 5 phones) and as im looking for a voip provider, i find that voip providers generally have unlimited plans, and those that offer sip origination and termination get charged for the minute, for their outgoing and incoming calls. </DIV> <DIV> </DIV> <DIV>is there a difference in the backend architecture here? if so, what? or is this is just a difference in marketing terms and setup?</DIV> <DIV> </DIV> <DIV>for example, <A onclick="return
top.js.OpenExtLink(window,event,this)" href="http://www.broadvoice.com/" target=_blank>http://www.broadvoice.com</A> offers an unlimited plan in the US for calls, though they never use the term sip origination and termination. they say their systems also supports asterisk. </DIV> <DIV> </DIV> <DIV>yet <A onclick="return top.js.OpenExtLink(window,event,this)" href="http://www.bandwidth.com/content/enterprise?page=voice_services_origination_termination&campaignId=701300000000JBJ" target=_blank>http://www.bandwidth.com/content/enterprise?page=voice_services_origination_termination&campaignId=701300000000JBJ </A> calls it sip origination and termination</DIV> <DIV> </DIV> <DIV>any info is appreciated! thanks!</DIV> <DIV> </DIV> <DIV> </DIV></DIV><BR>_______________________________________________<BR>--Bandwidth and Colocation provided by <A onclick="return top.js.OpenExtLink(window,event,this)" href="http://easynews.com/"
target=_blank>Easynews.com </A>--<BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> <A onclick="return top.js.OpenExtLink(window,event,this)" href="http://lists.digium.com/mailman/listinfo/asterisk-users" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR><BR><BR></BLOCKQUOTE></DIV><BR><BR clear=all><BR>-- <BR>Yair Hakak<BR>-----------------------------------------------------<BR>Yair Hakak, CEO<BR>Go Telecom, Ltd., Israel <BR>israel: (972) 54 5491266<BR>usa: (212) 202 2340<BR><A href="mailto:yair@gotel.co.il">yair@gotel.co.il</A> _______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE><BR>