Skip to Content.
Sympa Menu - Re: [] Re: Cisco 7960 and Avaya CCS 2.1 registration problem

Subject: SIP in higher education

List archive

Re: [] Re: Cisco 7960 and Avaya CCS 2.1 registration problem

Chronological Thread 
  • From: Stephen Kingham <>
  • To: Steve Blair <>
  • Cc:
  • Subject: Re: [] Re: Cisco 7960 and Avaya CCS 2.1 registration problem
  • Date: Mon, 31 Jan 2005 09:49:12 +1100


I would not call my additional comments as inspirational, but here they are

Steve Blair wrote:


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.

I am rather fond of the Cisco IOS based Gateways. These talk SIP on the LAN side and ISDN on the other.
eg 37xx, 26xx, AS5400. These all come with Primary Rate or Basic Rate ISDN interfaces - on the 26xx/37xx make sure yoiu get the interfaces with the Voice DSPs.

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.

Alternative is to use the Asterisk vmail-to-email.

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


Stephen Kingham wrote:


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 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 ;-(.




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


Stephen Kingham, MIT, BSc, E&C Cert
Project Manager and Consulting Engineer mailto:
Telephone +61 2 6222 3575 (office)
+61 419 417 471 (mobile)

Voice and Video over IP
for The Australian Academic Research Network (AARNet)

Archive powered by MHonArc 2.6.16.

Top of Page