asterisk users - Sep 2005 - MAX PRI for single server (was:Not enoughlinesavailable for Asterisk implemetation)

> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Troy Settle
> Sent: Wednesday, September 14, 2005 7:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] MAX PRI for single server (was:Not
> enoughlinesavailable for Asterisk implemetation)
> 
> I would be most interested in seeing some TNT/APX configurations and
> corrosponding SIP configurations for Asterisk.
www.voip-info.org - search for asterisk tnt
> 
> Right now, I'm using call routes and switching off a T1/PRI to my
> asterisk box, and would love to change that to pure SIP if possible.
> The only caveat is that my TNT boxes are primarily used for dialup
> traffic.
I have never tried a TNT for dual use, but it can be done. Might be too
much CPU load if there are a lot of calls.
> 
> Also, on the TNT, I see calling name information coming in from the
PSTN
> (Lucent 5E), but the TNT will not pass it through the PRI to my * box.
> Am I understanding correctly that calling name information also does
not
> work with SIP?
Calling name does work with SIP.

There is an issue with calling name delivery form a 5E to a TNT/APX if
the 5E is configured to do end office LIDB dips for calling name (like
qwest communications does it). The TNT does not understand the way that
the 5E sends "information following operation" and subsequent facility
IE containing the CNAM. BUT if you are seeing the CNAM on the TNT that
may not be the issues, if this problem is present you usually will not
see the name on the TNT either, just the number. 
> 
> Thanks,
> 
> --
>    Troy Settle
>    Pulaski Networks
>    866.477.5638
>    http://www.psknet.com
> 
> 
> 
> Damon Estep wrote:
> >  If you are looking for real high density VOIP termination I would
look
> > at
> >
> >>something like a Lucent APX 8000, configure correctly it can pass
> >
> > 2500+
> >
> >>g.729 calls to the PSTN course we paid lots of $ for ours.
> >>
> >>Chris
> >>
> >
> >
> >
> > Chris,
> >
> > My experience has been that the APX and TNT products require a
single
> > SIP proxy, how are you load balancing 2500 calls?
> >
> > If all of the traffic is outbound it is fine, but what about
> > origination? Are you using something other than asterisk as a SIP
proxy?
> >
> > On a smaller scale the TNT is a good bet since the number of calls
it
> > will do (672 with t3) is closer to what an asterisk box can do
without
> > trans-coding. You can connect 1 partially populated TNT to one * box
and
> > not need another sip proxy, you can also have a failover sip proxy
> > configured but not active unless the primary fails to respond.
> >
> > Both the TNT and APX have issues with calling name delivery over PRI
> > when connected to a Lucent 5ESS configured to do end office LIDB
dips,
> > so calling party name on inbound calls can be a bear, look to
connect to
> > a Nortel DMS if you have the option -- go figure the LUCENT media
> > gateways work better with Nortel class 5's than then they do with
lucent
> > class 5's.
> >
> > Have you learned something I have not about how to get all of the
calls
> > a TNT/APX can handle terminated on the SIP side without still having
a
> > single point of failure in the SIP proxy?
> >
> >
> >
> >
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