I would be most interested in seeing some TNT/APX configurations and
corrosponding SIP configurations for Asterisk.
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.
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?
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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