<div>Hi,</div>
<div> </div>
<div>The application is a IP-PBX with incoming calls through T1 and VOIP carriers. Asterisk will be used for Voicemail and IVR. I have written some AGI script to route calls based on user preference which is stored in the database. Users may connect through softphone or call in through PSTN. For example, a call comes in and depends on time/callerid/date, Asterisk will determine whether the call should be forwarded out, going to voicemail, etc.</div>
<div>For the purpose of SIP registering, I am thinking about using Openser/Freeswitch, whichever is more appropriate. If Freeswitch is more like a switch, should I be using it to replace Openser or part of the routing that is done by Asterisk?</div>
<div> </div>
<div>Thanks alot for your inputs.</div>
<div> </div>
<div>Pete</div>
<div> </div>
<div> </div>
<div><br><br> </div>
<div class="gmail_quote">On Fri, Apr 11, 2008 at 6:56 AM, David Knell <<a href="mailto:dave@3c.co.uk">dave@3c.co.uk</a>> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div text="#000000" bgcolor="#ffffff">Hi Pete,<br><br>Tell us a bit more about your application, and we'll be better able to help.<br><br>Cheers --<br><br>Dave<br>
<blockquote type="cite">
<div>
<div></div>
<div class="Wj3C7c">
<div>Hi all,</div>
<div> </div>
<div>I am currently evaluating either using Freeswitch or Openser for my SIP server. I am trying to look for information that can tell me the difference between the two in terms of ease-of-maintain, ease-of-implement, and scalability. If you have such kind of information, would you please share with me? </div>
<div> </div>
<div>Thank you very much for your input.</div>
<div> </div>
<div>Best Regards,</div>
<div>Pete</div></div></div><pre><hr width="90%" size="4">
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</pre></blockquote><br><br><pre cols="72">--
David Knell, Director, 3C Limited
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