I like the idea.<br><br>More and more SIP trunks will be turning up. Why bother having to go from H323 to SIP. Simpler just to run SIP.<br><br>I also like SIP and how you can set it up to monitor the destination of your dial-peers. Shut them down if a CCM is down.<br>
<br>Cheers,<br><br>Tim<br><br><div class="gmail_quote">On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <span dir="ltr"><<a href="mailto:SCASPER@mtb.com">SCASPER@mtb.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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<div>I assume you are talking traditional analog and digital PSTN gateways, why are you considering migrating to SIP to control these as opposed to H323? .</div>
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<div>Steve<br><br>>>> Voice Noob <<a href="mailto:voicenoob@gmail.com" target="_blank">voicenoob@gmail.com</a>> 11/3/2009 6:09 PM >>><div class="im"><br>Has anyone started using SIP on the PSTN gateway? I want to use it instead of H.323 or MGCP and start migrating it to SIP on the gateway. Any experience with this? Can I get Calling Name and Number from the PSTN side? </div>
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<br></blockquote></div><br><br clear="all"><br>-- <br><br>Cheers,<br><br>Tim<br><br><br>Sent from Sydney, Nsw, Australia