[asterisk-users] Re: Question about DSP in Digium card

[A. Levy]

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 at 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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