SIP in higher education

[Steve Blair <>]

Stephen:

 Thanks. I get it. I'm asking because I have a bit of a twisted environment
with an Octel voicemail system that currently is only accessible via
a PRI to Verizon. I have a few conflicting requirments for voicemail
for SER users and I am looking for inspiration. So....

  My Octel is setup for SMDI signaling. If I want to provide access from
SER without traversing Verizon's DMS100 then I need a gateway. The
only gateways I know about speak MGCP not SIP on the LAN side.
This worked with our Call Manager trial because CM spoke MGCP.

 Attempts to get MGCP and SMDI working on my Asterisk server
have been unsuccessful. If all else fails I can do rewriting to get the
call to Octel via Verizon but I need to do the rewrite before applying
the CC-Diversion header. If I wait until the header is applied then the
originally called number (OCN) sent in the ISDN IE to Verizon
contains an insufficient number of digits.

  Like I said all of this is workable but I am looking for inspiration. :-)

Thanks,Steve

Stephen Kingham wrote:

> Hi
>
> Dennis Baron wrote:
>
> > >               Is this the configuration you use with your SER proxy?
> > > If so do your IP phone users have the ability to do extension dialing
> > > between themselves and/or other on campus non-IP phones? If they
> > > do how many digits are in your extensions and how did you implement
> > > this in SER?
> > >   
> >
> >
> > We're running SIPxchange and not SER.  But yes, SIP accounts all have
> > a five digit alias, and usually a SIP.edu email alias, for the ten
> > digit SIP user.  The dial plan supports both five digit dialing to the
> > PBX and 9+ dialing to the PSTN via the PBX.  Plus just about
> > everything else that you can dial on the PBX.
> >  
>
> I support the same on SER.  I do that by using rewrite rules in the 
> ser.cfg.
>
> At the start of processing any INVITES I check if it is an E.164.  If 
> it is I convert the various Dial Plans into the full international 
> DialPlan including the +.  Then as I process ACL (long distance 
> baring), ALIAS table, and Routing number ranges to various Gateways I 
> only deal with one dial plan.  The user can think they are at home, on 
> a PBX extension, etc, and use the Dial Plan they are use to, 
> internally my routing and barring etc only deals with one dialplan.
>
> Then optionally for some users I convert using the ALIAS table  all 
> the contacts for a user (including calls to their hard phone and PBX 
> phone and Cell phone numbers) to their SER username.  I Then use 
> static entries in the Location Table to send calls to that user's PBX 
> extn, Cell Phone, and of course any UAs.  So the user can have 
> parallel forking.
>
> PS This stuff is hard to describe with just text ;-(.
>
> Stephen
>
> >              Dennis
> >
> > ========================================================================= 
> >
> >
> > Dennis Baron; Information Services & Technology     
> > mailto:
> > Senior Strategist for Integrated Communications        
> > sip:
> > Massachusetts Institute of Technology; Room E19-738   
> > tel:+1-617-252-1232
> > 77 Massachusetts Avenue;  Cambridge, MA  02139-4307   
> > fax:+1-617-253-8000
> >
> > ========================================================================= 
> >
> >
> >  

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