I don't know anyone chime in?<br><br>
<div class="gmail_quote">On Tue, Nov 3, 2009 at 5:17 PM, Daniel <span dir="ltr"><<a href="mailto:dan.voip@danofive.id.au">dan.voip@danofive.id.au</a>></span> wrote:<br>
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<div>Is there limitations to DTMF and requiring MTP to pass it through?</div>
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<div>That's what I would be looking at I know we had similar issues with MPX as a SIP trunk.<br><br></div>
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<div class="im">On Wed, Nov 4, 2009 at 9:09 AM, Voice Noob <span dir="ltr"><<a href="mailto:voicenoob@gmail.com" target="_blank">voicenoob@gmail.com</a>></span> wrote:<br></div>
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<div class="im">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? <br>
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