Chris,<br>
<br>
See my inline comments.<br>
<br>
Regards,<br>
Paul<br><br><div><span class="gmail_quote">On 4/22/05, <b class="gmail_sendername">Chris St Denis</b> &lt;<a href="mailto:chris@aebc.com">chris@aebc.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I have some questions about SER that I can't find answers to in the wiki.<br><br>Does SER have T.38 support?</blockquote><div><br>
SER technically is oblivious to multimedia streams, however, if your
SIP UA supports fax protocols then it should work. If you're having to
deal with NATed SIP UAs then you're still in luck because the NAT
traversal &quot;helpers&quot; (ie, mediaproxy or rtpproxy, which are both
__external__ to SER) do support T.38.<br>
</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Does SER have a module to send MWI notifications?</blockquote><div><br>
MWI is simply a SIP message type called NOTIFY. SER, however will not
ever generate a NOTIFY message because it is not a voice mail system.
Many people use Asterisk as a voicemail server, which will generate a
NOTIFY message and SER can proxy it to the actual SIP UA. So the short
answer is yes?<br>
</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Does the SER nat support replace a STUN server? Should it be used with one?<br>Is stun better to use than it?
</blockquote><div><br>
IMHO, you should avoice STUN when possible. We have successfully used
mediaproxy and rtpproxy for transparent NAT traversal. So the SIP UA
doesn't have to do squat to work. Visit <a href="http://onsip.org/">http://onsip.org/</a> to get the
Getting Started document which fully describes setting up both rtpproxy
and mediaproxy.<br>
</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Can SER handel more than one database type loaded? (Eg mysql and pgsql?) If<br>so, how can you choose which one is used for what?
</blockquote><div><br>
I don't know - but I suspect you should only use one or the other.<br>
</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Any (Unix) ODBC database module planned or under development?</blockquote><div><br>

I don't think so. SER uses native DB connectivity so be as fast as possible.<br>
</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">When routing a call through ser (from PSTN or another sip phone) how can I<br>define the number of rings before the call gets sent to voicemail?
</blockquote><div><br>
The SER tm module has a timer&nbsp; value called&nbsp; fr_invite. You
can search the mailing list archives for dynamic invite timers to get a
hint.<br>
</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">_______________________________________________<br>Serusers mailing list<br><a href="mailto:serusers@lists.iptel.org">
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