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

2007-03-28 Thread Matthew Fredrickson


On Mar 27, 2007, at 8:35 AM, Salvatore Giudice wrote:
As for the DSP, you are right to be concerned about the Digium cards, 
but not because of the DSP. The DSP is not where you will run into 
problems. Digium cards feature 2 year old circuitry and do not play 
well with other devices. You have to take care not to share interrupts 
with any components that may be active on that system. Sharing an IRQ 
between a Digum card and an Ethernet card would certainly spell 
disaster in my experience.

 
From personal experience, I no longer use Digium hardware since I 
could rarely push a quad port card to more than 13 channels per T1 
circuit without the card failing miserably. HDLC aborts abound.


FWIW, there have been some recent improvements in the drivers and 
firmware which correct most of the old IRQ sharing and HDLC problems of 
that nature.  If you have any more such problems, be sure to let tech 
support know so we can get it fixed.  We are anxious to keep your 
business.


Matthew Fredrickson

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

2007-03-27 Thread Salvatore Giudice
 

 

You've got a decent server. Generally the limiting factor for the number of
simultaneous calls is more about server memory. That server could likely
handle 124 simultaneous calls, but you would be prudent to double that
memory size. Make sure you are running at 100 full especially if you are
using G.711. 10 Full uplinks won't cut it if you are running that kind of
bandwidth.

 

As for the DSP, you are right to be concerned about the Digium cards, but
not because of the DSP. The DSP is not where you will run into problems.
Digium cards feature 2 year old circuitry and do not play well with other
devices. You have to take care not to share interrupts with any components
that may be active on that system. Sharing an IRQ between a Digum card and
an Ethernet card would certainly spell disaster in my experience.

 

>From personal experience, I no longer use Digium hardware since I could
rarely push a quad port card to more than 13 channels per T1 circuit without
the card failing miserably. HDLC aborts abound.

 

For now, I only use Sangoma cards. These don't have the IRQ issues and I
have had no problems pushing their cards to their maximum. I recommend echo
canceller enabled cards for any T1/E1's you may use that are not long
distance carrier lines. 

 

Good luck, hope this helps with your capacity planning. - SG

 

 

 

 

 

 

 

 

##

2007/3/24, A. Levy <[EMAIL PROTECTED]>: 

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

2007-03-27 Thread Brad Sumrall
Whether it is IAX, SIP, H323 or ?

 

These are authentication handshakes to establish an rtp stream.

 

SIP = user name and password in a standardized IP packet

IAX = same

H.323 = same

 

Is also has to do with what codec are supported as well.

 

As far as NAT is concerned!

 

Yep, tell your ISP to forward the authentication port or just junk their
gear and get something like a low end Cisco.

 

Or

 

Get IP Phones with STUN (a little pricey)

 

Or

 

Trick

 

Use some type of tunneling gear to an outside IP (outside your NAT) and then
bounce your authentication from this new gateway!!!

i.e. establish a VPN connection to an outside router from an internal router
and drive the call through there.

 

Brad

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of A. Levy
Sent: Tuesday, March 27, 2007 6:54 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Question about DSP in Digium card

 

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 <[EMAIL PROTECTED]>: 

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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