[asterisk-users] DISA and legacy PBX

Colin Anderson ColinA at landmarkmasterbuilder.com
Wed Oct 4 07:49:16 MST 2006


I've used the prompt pls-wait-connect-call to give my users a cue to cool
their heels for a second or two in circumstances like this, and no one has
complained. That's probably the most useful prompt in Asterisk! 

-----Original Message-----
From: James Harper [mailto:james.harper at bendigoit.com.au]
Sent: Wednesday, October 04, 2006 1:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DISA and legacy PBX


I've configured our PBX so that when a user dials 80 on the PBX
extension, it goes out an ISDN TE interface on the PBX and into an NT
interface on my asterisk machine, where it jumps into the 's' extension.

Asterisk then does a DISA(no-password|sip_provider_out) which allows the
call to go out via a sip provider, to give us cheaper calls.
Unfortunately if the user doesn't wait for DISA to give dialtone,
asterisk doesn't hear all of the digits.

When they dial '0' on the PBX there is no need to wait for dialtone, so
it is a bit confusing for the users.

Any suggestions? I'm using misdn, and 'immediate' and 'always_immediate'
in the config for that port, so maybe there is something I could do
there to take whatever digits have been dialled so far...

Thanks

James
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