[Asterisk-Users] inbound call routing problem

[C. Maj]

On Tue, 13 Jan 2004, Lane Hoskins waxed:

> We have 8 lines coming into an ADTRAN channelbank that then goes to the
> * server via a T100P card. I need to route lines 1 and 2 to everyone
> when a call comes in on either of them. I also need lines 3 - 8 to ring
> first at specific sip extensions (direct dials for staff here) and then
> to go to voicemail or fwd to a cellphone after that if the extension is
> not answered.  Has anyone done this that could provide an example for me
> or point me to better documentation? We have searched extensively and
> not found anything yet.

Here's Rich Adamson's "A WORKING EXAMPLE" from September:

http://lists.digium.com/pipermail/asterisk-users/2003-September/020944.html

I see SIP and Voicemail in there, but I haven't tried it
myself.

--Chris


-- 

Chris Maj <cmaj_hat_freedomcorpse_hot_info>
Pronunciation Guide:  Maj == May
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