[Asterisk-Dev] DISA

Ed Devine ncfm at airmail.net
Thu Feb 5 17:02:29 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.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20040205/6bd391b6/attachment.htm

More information about the asterisk-dev mailing list