Asterisk and Diguim Card

I just want to know that can i use two diguim card at a time in asterisk . if yes then how can i differentiate b/w two cards , and how to set the span numbers of those two card in zaptel.conf .

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Googling, I found this:

The thread indicates that it will work, but each card has to be on its own IRQ.
Yes you can install two cards although that is the recommended maximum since they generate a lot of interrupts and having more than 2 cards can cause issues.

As far as how you differentiate between the cards and configure the span number it depends on what cards they are. For example if the two cards are a different type such as a 4 port analogue interface and a ISDN E1 line then it will depend on the order in which the two zaptel drivers are loaded. If you load the analogue driver first then spans 1-4 will be the analogue interface while 5-35 (30 channels plus the signaling D channel) will be for the ISDN.
If both cards are identical then it depends which slot the cards are placed in the motherboard.
prince910Author Commented:
Thankz for the reply ... okie i have two same type of diguim card i.e TE420B.

> So , as u describe above if i have two same card like i have then, in this case the spans will be 1-8 ?
> These two cards can operate simultaneously ?
> I also have two gateways i.e Mediant 1000 with 4 E1/T1 per gateway and those two gateways are going to be sync with those diguim TE420B cards.
> And for the signalling point of view b/w Mediant 1000 and the diguim card should i use signalling pri_cpe and pri_net?
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Its a bit different with PRI interfaces.
You will have one span per port.
Each port with have 23 (T1) or 30 (E1) data channels and one signalling channel. E1 is european and can carry 30 calls while T1 is USA and can carry 23.
You then group the 8 spans into two groups so that when dialing you choose the group and therefore card in which to dial out of.

One end of the PRI needs to use PRI_NET (behaves like a telco) and the other like PRI_CPE (like a telephone system in customer premesis). It might be easier to get asterisk to use PRI_NET as it can definetly do it.

If you let me know what country you are in and whether you intend to use E1 or T1 then I can give you some configuration examples. PRI also uses different signalling types which vary between countries.
prince910Author Commented:
I am not using National Pri ... i want to sync the medaint 1000 voice gateway with my digium card.
prince910Author Commented:
and thaknzzz for ur reply it gives me alot of help in configuration. and its clear my all doubts that i want to know. Also one more thing did Linux distribution Centos5 support for SAS drive drivers ???
SAS doesn't have a single driver unlike IDE. SAS drive support will depend on what SAS controller is being used.
prince910Author Commented:
Right now i have no problem with SAS driver as i have overcome on that ... Now i install the TE420B card and i want to know how  can I up that module ? and which module is in zaptel for TE420B as there are others like wctc4xxp,wct4xxp etc but is there n e particular module to be loaded for TE420B bcz i am unable to load that .... But it is installed on the system as i do lspci

"11:08.0 Communication controller: Digium, Inc. Unknown device 0420 (rev 02)"

Above 0420 is the identifer fo the TE420B card. but what will be the selectable modlule in zaptel for that card ?
You should be using the wct4xxp driver.

Make sure you are running the latest copy of zaptel. Support for these cards was included in the source on the 5th of July.
prince910Author Commented:
Thankzz grblades ... Now my server is up and running with two TE420B card simultaneously. Also i have VICI dialer installed on it ,

> first question is related to the outbound trunk in extension.conf as i am using the digium card to dial out and i have total 8 ports so after each 30 channel i have to shift to the next E1 slot and for that i have to set GOTOIF statements in extensions.conf like

;exten => _1XXXXXXXXXX,2,Dial(SIP/${EXTEN}@asghar,50,tT)
;exten => _1XXXXXXXXXX,3,Gotoif($[${DIALSTATUS} = CHANUNAVAIL]?7)
;exten => _1XXXXXXXXXX,4,Gotoif($[${DIALSTATUS} = CONGESTION]?7)
;exten => _1XXXXXXXXXX,5,GoToIf($[${DIALSTATUS} = BUSY]?7)
;exten => _1XXXXXXXXXX,6,GoToIf($[${DIALSTATUS} = HANGUP]?7)
;exten => _1XXXXXXXXXX,7,Dial(Zap/g3/${EXTEN},50,tT)
;exten => _1XXXXXXXXXX,8,Gotoif($[${DIALSTATUS} = CHANUNAVAIL]?2)
;exten => _1XXXXXXXXXX,9,Gotoif($[${DIALSTATUS} = CONGESTION]?2)
;exten => _1XXXXXXXXXX,10,GoToIf($[${DIALSTATUS} = BUSY]?2)
;exten => _1.,10,Dial(SIP/2692${EXTEN}@richmedium3)
..... and so on

> as i am using the VICI dialer .. so is there n e other way in VICI dialer so that i can set different trunks for range of agents so that i dun have to do the GOTOIF stuff.
In your example above as you are dialing using SIP and then if that fails dial using ZAP then yes you do have to use the gotoif on the dialstatus.

However there is no need to shift to the next E1 slot if there is an error as you can define a group which just has all the channels in it.

For example in your zapata.conf file you probably have something like :-

switchtype = euroisdn
signalling = pri_cpe
group = 1
channel => 1-15
channel => 17-31
group = 2
channel => 32-46
channel => 48-62

You could change this by just removing the additional 'group = 2 ... 8' lines so that all the channels are in the same group.

prince910Author Commented:
if i do as u said like overlap the group and just create only one group .
then what would be dial plan for that .. in dial plan i just have to use only one group i.e group=1 or what ?
Yes in the dial plan you just use group 1 for all calls.
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