[cisco-nas] AS5350XM in mixed VoIP and dialup environment

[Darryl Sladden (dsladden)]

Bernhard,

There are many questions there, and some, such 
as the number of DSP in use, may be better handled 
by a call to TAC.

First, you need to have a separate Voice DSP license 
before you can proceed with accepting VoIP calls on 
the AS5350.

Second, the method that you would use for VoIP calls 
would be based on dialpeers.  The dialpeer matching 
would occur BEFORE the Group-Async command was used.

The configuration for a VoIP GW is VERY different then
a standard NAS box.

You would generally use dialpeers and not dialer interfaces.
Dialer does not apply at all to VoIP calls.

Calls can be TDM hairpinned from the E1 connect to the 
PSTN to the E1 connected to the channel box and not 
use DSP ONLY IF They are ISDN signalled Voice calls.
(ie CAS calls that have inband signals require a DSP.) 

This architecture is well deployed and you should not 
have any major problems, but you many need to upgrade 
if you reach capacity issues.

Regards,
Darryl Sladden
Product Manager AS5000
Cisco Systems - ABU
dsladden at cisco.com
408-525-8970 

 

> -----Original Message-----
> From: cisco-nas-bounces at puck.nether.net 
> [mailto:cisco-nas-bounces at puck.nether.net] On Behalf Of 
> Bernhard Schmidt
> Sent: Tuesday, January 03, 2006 10:51 AM
> To: cisco-nas at puck.nether.net
> Subject: [cisco-nas] AS5350XM in mixed VoIP and dialup environment
> 
> Hi everyone,
> 
> I sent some of those questions to cisco-nsp already, but 
> maybe this (or at least for some parts cisco-voip) is the 
> better forum.
> 
> we're currently evaluating two 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. 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.
> 
> 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 second question is about the use of the SPE dsps. We 
> patched some lesser used lines/numbers to the two Ciscos we 
> got and the output of "sh spe" does not look to good. E.g. at 
> the moment we have 5 modem users (no compression on the 
> group-async and no multilink) and  6 digital users (multilink 
> and compression allowed), of which three are ordinary 
> dialups, two have multilink enabled but only one channel in 
> use and one has two channels. Still, show spe shows
> 
> Ports     : Total  216  In-use    61  Free   155  Disabled     0
> Calls     : Modem    5  Digital    0  Voice    0  Fax-relay    0
> 
> the box has two NP108 and one 8E1, but if it uses 61 DSP 
> lines on 11 calls we won't be remotely able to fill the 
> lines. Does active just mean "powered because it was used at 
> some time" or is it used to an incoming call can't use it anymore?
> 
> We run IOS 12.4(1c) as preinstalled on the boxes.
> 
> Regards,
> Bernhard
> 
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