[Asterisk-Users] Re: [Asterisk-Dev] DISA

John Todd jtodd at loligo.com
Thu Feb 5 18:17:31 MST 2004


>Hi All!
>
>Okay, let me understand this. No offense intended. I've struggled 
>for about a month with an Asterisk system, seeking to establish at 
>least the minimal functionality of the PBX we wanted to retire 
>(Nortel SL1). My objective was to try and use Asterisk as a 
>replacement/backup telephone switch.
>
>Although no one I've spoken with has said as much, it appears (based 
>on my nearly constant efforts and the reems of downloaded code I've 
>gone through) that the Asterisk application lack's the one 
>capability that a large segment of the telephone market (CLEC's, as 
>in my case, but virtually all service providers, etc...) require.
>
>Apparently,  Asterisk doesn't really work like a traditional pbx in 
>that you really can't (for example) select a line (say from a 
>Norstar using a T1 connected to a Digium T410p) go off hook and get 
>dialtone.
>
>Nor (and I understand the security issues of a DISA environment) 
>does it appear that users can readily dial into an Asterisk system, 
>get dialtone, and dial a call.
>
>I've reviewed, massaged, monkeyed with app_disa.c, and while it is a 
>well done and serviceable application, it lacks the flexibility 
>necessary to adequately address real world uses.
>
>Anyway, before I trash the project entirely and sell the equipment, 
>I wanted to make sure that I really inderstood that Asterisk isn't 
>(at present) capable of  volume call switching in a DISA application.
>
>

[moved to asterisk-users, as it is more on-topic there]

I would disagree with your summary.

Asterisk does not work well in an environment where you're connecting 
two lines together without dialing anything, though I can't say that 
I've tried this:

exten => 1234,1,Dial(Zap/1-2/w)

Where Zap 1-2 was a non-PRI T1 channel that had "dialtone" on it. 
Perhaps that would work.  But that method would be foolish and 
somewhat crippled in functionality.

But, to your argument, you certainly can create dialplan rules that 
just connect one line to another.  If you have given the description 
above to people who said that it could not be done, then I suspect 
you have talked to people who have not done it and who were only 
marginally clued in as to how Asterisk works.

JT



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