[c-nsp] AS5350XM in mixed VoIP and dialup environment

[Bernhard Schmidt]

Hi everyone,

we're currently evaluating a Cisco AS5350XM for use in our university
network. It should replace the old Ascend TNT boxes for ISDN/modem
dialup and provide a SIP/PSTN gateway to use for our VoIP PBX to be
installed next year. I don't have that much experience with Cisco Dialup
(although that part seems to work)and I'm highly confused now with the
whole dial-peer thing. I hope someone can shed some light on my
misunderstandings, maybe I'll get it some day... :-\

The whole setup will basically look like this

+----------------+
|      PSTN      |
+----------------+
  | | | | | | 
  | | | | | | 6*E1
  | | | | | |
+----------------+     E1    +-------------+
|    AS5350XM    |-----------| channel box |
+----------------+           +-------------+
        +
	+ SIP
	+
+----------------+
|   SIP PBX (*)  |
+----------------+

Current setup is just one E1 to our HiCom.

All six E1 lines will be configured the same way and will have a large
block as well as several additional numbers configured (so calls to one
number can be signalled on one random E1). 

Data (ISDN) or Voice (modem) calls to several numbers on that trunk
should be handled by the box itself (ordinary PPP dialup). A small block
of our numbers should be sent to the channel box (so that is basically
PSTN-to-PSTN switching). An important thing here would be that PSTN to
the channel box is transparent regarding data, so we can connect any
device there. All remaining destinations should be sent to the PBX with
SIP.

The other way around calls coming from the SIP PBX should be sent to
PSTN. Reaching the channel box and the dialin would be great, but isn't
necessary. Same for the channel box, reaching PSTN is the important
thing, dialin and PBX would be of additional value.

So far during tests we managed to get several things working
independently:

* ISDN and modem dialup with PPP on any number sent to the AS (so no
  number matching at all)
* PSTN to SIP gateway
* SIP to PSTN gateway

After heavy fiddling (especially with the dial-peers) I got those
running, but I still don't get the big picture so its more or less
guesswork.

My first question is regarding dialup. Currently we have the whole
dialup configuration on Serial3/0:15 and additionally on Group-Async0,
which has both 108port spes configured in (group-range 1/00 2/107). This
makes the Cisco answer each and every call it receives with PPP. If I
wanted to connect to different "configuration profiles" depending on the
dialed number, I had to put a "dialer pool-member x" on the lines and
use it in several Dialer-interfaces with the same "dialer pool x" and
different "dialer called <number>", correct? Would that work with async
(modem) connections as well? As far as I understood the documentation it
won't, and I would need to use resource-pooling to assign specific
numbers to specific async lines and then group those lines specifically.

The next question is about call routing. I've managed to get calls
working between PSTN and the SIP PBX with the following dial-peer
configuration:

dial-peer voice 4 pots
 translation-profile incoming FROM-PSTN
 translation-profile outgoing TO-PSTN
 destination-pattern ^3333
 incoming called-number .
 direct-inward-dial
 port 3/0:D

dial-peer voice 5 voip
 translation-profile incoming FROM-SIP
 translation-profile outgoing TO-SIP
 service session
 destination-pattern ^2222
 session protocol sipv2
 session target ipv4:129.187.10.22
 session transport udp
 incoming called-number .

FROM-PSTN adds 2222 to the number, TO-SIP strips it again. FROM-SIP adds
3333 to the number, TO-PSTN strips it. This way we make sure that
every call is switched to the "other side" of the AS5350 to avoid
configuring every extension at the (at the moment temporary) SIP PBX on
the router. This seems to work fine, although I still fail to get the
significance of "incoming called-number .".

Anyway, I wanted to terminate one number (27891 on PSTN) on dialup now.
After some searching on CCO I got the statement

dial-peer voice 1 pots
 service data_dialpeer
 destination-pattern 222227891

but regardless of what I enter in the destination pattern (^222227891,
27891), this dial-peer is never matched and my modem calls are handed
off to the PBX. I even tried to exclude this particular number in
FROM-PSTN to avoid getting a 2222 prefix in front of it and tried to
match to destination-pattern 27891. I got a modem connection but only
realized later that none of my dial-peers had matched and modem was
default.

With destination-pattern 27891 "show dialplan number 27891" says "No
match, result=-1". Somehow I don't get the whole concept of inbound leg
matching, outbound leg matching and which commands have to match where
to have a successful call. And how will that interfere with the "dialer
called" thing above anyway?

I'm really scared of having the channel box configured into this mess at
the moment as I don't understand the concepts. I couldn't find a good
documentation yet, is there any particular URL or book one should have
read before attempting to configure this?

Thanks in advance,
Bernhard