Chris,

See my inline comments.

Regards, Paul

On 4/22/05, Chris St Denis chris@aebc.com wrote:

I have some questions about SER that I can't find answers to in the wiki.

Does SER have T.38 support?

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 "helpers" (ie, mediaproxy or rtpproxy, which are both __external__ to SER) do support T.38.

Does SER have a module to send MWI notifications?

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?

Does the SER nat support replace a STUN server? Should it be used with one?

Is stun better to use than it?

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 http://onsip.org/ to get the Getting Started document which fully describes setting up both rtpproxy and mediaproxy.

Can SER handel more than one database type loaded? (Eg mysql and pgsql?) If

so, how can you choose which one is used for what?

I don't know - but I suspect you should only use one or the other.

Any (Unix) ODBC database module planned or under development?

I don't think so. SER uses native DB connectivity so be as fast as possible.

When routing a call through ser (from PSTN or another sip phone) how can I

define the number of rings before the call gets sent to voicemail?

The SER tm module has a timer value called fr_invite. You can search the mailing list archives for dynamic invite timers to get a hint.

_______________________________________________

Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers