asterisk users - Mar 2007 - Question about DSP in Digium card

Hello.

I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find
out if there is any limitation about DSP capabilities, I mean, I am not sure
how many phone calls my Digium card supports, simultaneously. The calling
flow goes from IAX <-> ISDN.

I am running this card into CPU like this:
- Micro PIV 3.0
- 1Gbyte Memory


Thanks.

Levy.-
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A. Levy wrote:
> Hello.
> I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to 
> find out if there is any limitation about DSP capabilities, I mean, I 
> am not sure how many phone calls my Digium card supports, 
> simultaneously. The calling flow goes from IAX <-> ISDN.
> I am running this card into CPU like this:
> - Micro PIV 3.0
> - 1Gbyte Memory
> Thanks.
> Levy.-
120 simultaneous calls.  AFAIK the DSP functionality is mostly offloaded 
to the the server's CPU so the server is going to be your bottleneck, 
not the card.  Your system should be OK I think.  I ran 95 simultaneous 
call through HP DL320's with 3ghz P4 and a gig of RAM and straight 
ULAW.  Top showed 50%-60% utilization with all 95 channels in use.  
These boxes only served as gateways between TDM and SIP and passed the 
calls to another Asterisk server that handled all of the other 
functionality (queues, meetme, etc)

Just my personal experience, but I do not find IAX to be very reliable.  
Is there any particular reason you are not using SIP?

Thanks,
Steve Totaro
www.asteriskhelpdesk.com

well, ...,we did not choose SIP because our customers are located behind NAT
router (using private IP's) and those routers
are not managed by them but by the ISP so it is very difficult to establish
full duplex phone calls because
you can not initiate voice over ip session from the internet (outside) to
LAN side (inside) with private IP's. We could not establish
2-way phone calls, I mean, the conversation is listened in 1-way only. As I
mentioned before, we can not configure PAT into the NAT router neither
because is handled by the ISP and the passwords are unknown ....
That's  why we decided to use IAX instead of SIP, I mean, IAX is more robust
than SIP when the NAT router is 3th-party managed and
the PAT feature is not enable.
On the other and we tested IAX over dialup links and it worked fine....
Those are the reasons we choose IAX as "acess protocol" to our
SIP/H323
Network. You know, the access networks of the customers are different
completely: Private IP Address over DSL lines (NAT Router), Public IP
Address over DSL lines, Corporate Networks over dedicated Links (Public
and IP Addresses), Dialup links, ......
Any comment would be welcomed,
thanks a lot

Levy.-

2007/3/24, A. Levy <ardielle@gmail.com>:
>
> Hello.
>
> I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to
find
> out if there is any limitation about DSP capabilities, I mean, I am not
sure
> how many phone calls my Digium card supports, simultaneously. The calling
> flow goes from IAX <-> ISDN.
>
> I am running this card into CPU like this:
> - Micro PIV 3.0
> - 1Gbyte Memory
>
>
> Thanks.
>
> Levy.-
>
>
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