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<p>SIP/H323 - PSTN - USA and Canada</p>
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<div>Wow, a VoIP termination service with Asterisk. <br>That's amazing. <br>Congratulations.<br><br>><i> -----Original Message-----<br></i>><i> From: <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">
asterisk-biz-bounces at lists.digium.com</a><br></i>><i> [mailto:<a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">asterisk-biz-bounces at lists.digium.com</a>]On Behalf Of jltaylor<br></i>><i> Sent: 04 May 2005 00:55
<br></i>><i> To: Commercial and Business-Oriented Asterisk Discussion<br></i>><i> Subject: RE: [Asterisk-biz] $0.006 USA/Canada Termination<br></i>><i> <br></i>><i> <br></i>><i> After much work, study, reading, collaborating, and una mordita, I have
<br></i>><i> successfully built a termination service with Asterisk.<br></i>><i> It works, I use it every day.<br></i>><i> PRI->asterisk->VOIP<br></i>><i> PRI->asterisk->VOIP->PRI(PSTN)<br></i>>
<i> <br></i>><i> James<br></i>><i> <br></i>><i> -----Original Message-----<br></i>><i> From: <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">asterisk-biz-bounces at lists.digium.com</a><br></i>
><i> [mailto:<a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">asterisk-biz-bounces at lists.digium.com</a>]On Behalf Of Ed Kenny<br></i>><i> Sent: Tuesday, May 03, 2005 3:38 PM<br></i>><i> To: 'Commercial and Business-Oriented Asterisk Discussion'
<br></i>><i> Subject: RE: [Asterisk-biz] $0.006 USA/Canada Termination<br></i>><i> <br></i>><i> <br></i>><i> "So you want to route a call (that came into your * box via your PRI) to<br></i>><i> a VOIP endpoint or gateway."
<br></i>><i> That's correct. I know how to route a call but not terminate a call.<br></i>><i> There was an earlier post here suggesting VOIP termination via Asterisk<br></i>><i> was easy. I'm not sure I believe that. If $0.006 is really available
<br></i>><i> from viable third party services then my choice of whether to build or<br></i>><i> outsource is in the air.<br></i>><i> <br></i>><i> So specifically for this forum are the questions of who are the viable,
<br></i>><i> low cost VOIP terminators?<br></i>><i> And, has anyone else here successfully built there own termination<br></i>><i> service with Asterisk?<br></i>><i> <br></i>><i> I'll save the "how to" build such for the user forum.
<br></i>><i> <br></i>><i> Thanks<br></i>><i> <br></i>><i> <br></i>><i> -----Original Message-----<br></i>><i> From: <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">asterisk-biz-bounces at lists.digium.com
</a><br></i>><i> [mailto:<a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">asterisk-biz-bounces at lists.digium.com</a>] On Behalf Of Rusty<br></i>><i> Shackleford<br></i>><i> Sent: Tuesday, May 03, 2005 3:47 PM
<br></i>><i> To: 'Commercial and Business-Oriented Asterisk Discussion'<br></i>><i> Subject: RE: [Asterisk-biz] $0.006 USA/Canada Termination<br></i>><i> <br></i>><i> > -----Original Message-----<br></i>>
<i> > From: <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">asterisk-biz-bounces at lists.digium.com</a><br></i>><i> > [mailto:<a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">asterisk-biz-bounces at lists.digium.com
</a>] On Behalf Of Ed Kenny<br></i>><i> > Sent: Tuesday, May 03, 2005 10:12 AM<br></i>><i> > To: 'Commercial and Business-Oriented Asterisk Discussion'<br></i>><i> > Subject: RE: [Asterisk-biz] $0.006 USA/Canada Termination
<br></i>><i> ><br></i>><i> ><br></i>><i> ><br></i>><i> > OK, let me try to make my primary question more obvious.<br></i>><i> ><br></i>><i> > How would an Asterisk host of inbound toll free PRI calls
<br></i>><i> > allow these callers to dial direct long distance calls via<br></i>><i> > the Asterisk host's Internet bandwidth (VOIP), thereby not<br></i>><i> > utilizing a PRI channel for the outbound call?
<br></i>><i> <br></i>><i> So you want to route a call (that came into your * box via your PRI) to<br></i>><i> a VOIP endpoint or gateway.<br></i>><i> <br></i>><i> That's a very basic operation that is well documented in the available
<br></i>><i> resources. Beyond mentioning that you are probably going to want to shop<br></i>><i> for VOIP "termination", such a discussion is better suited to the<br></i>><i> "users" list, after you've availed yourself of the aforementioned
<br></i>><i> documentation.<br></i>><i> <br></i>><i> --<br></i>><i> No virus found in this outgoing message.<br></i>><i> Checked by AVG Anti-Virus.<br></i>><i> Version: 7.0.308 / Virus Database: 266.11.2
- Release Date: 05/02/2005<br></i>><i> <br></i>><i> <br></i>><i> _______________________________________________<br></i>><i> Asterisk-Biz mailing list<br></i>><i> <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">
Asterisk-Biz at lists.digium.com</a><br></i>><i> <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">http://lists.digium.com/mailman/listinfo/asterisk-biz</a><br></i>><i> <br></i>><i> _______________________________________________
<br></i>><i> Asterisk-Biz mailing list<br></i>><i> <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">Asterisk-Biz at lists.digium.com</a><br></i>><i> <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">
http://lists.digium.com/mailman/listinfo/asterisk-biz</a><br></i>><i> <br></i>><i> _______________________________________________<br></i>><i> Asterisk-Biz mailing list<br></i>><i> <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">
Asterisk-Biz at lists.digium.com</a><br></i>><i> <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">http://lists.digium.com/mailman/listinfo/asterisk-biz</a><br></i><br clear="all"><br>-- <br>VoIPInvite<br>
3888 Duke of York Blvd, Suite# 1424<br>Mississauga, ON, L5B 4P5<br>Canada<br>P 1 416 828 5262<br>F 1 360 483 2170<br><a href="mailto:vijay.shan@voipinvite.com">vijay.shan@voipinvite.com</a><br><a href="http://www.voipinvite.com">
www.voipinvite.com</a> </div>