[Asterisk-Users] Asterisk - first impressions

[cisb]

REMOVE
----- Original Message -----
From: "Peter Zeltins" <pzeltins at bigfoot.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, June 26, 2003 3:16 AM
Subject: [Asterisk-Users] Asterisk - first impressions


> I'm still a newbie in Asterisk, just yesterday installed it for home use
(so
> I can call home while travelling). Using AVM A1 (BT Speedway) ISDN card.
> Anyway, I find it very hard to locate supporting information for Asterisk.
> User's Handbook is still a draft, this mailing list is not easily
> searchable, and any info out on Usenet is scarce. I spent some 30mins just
> trying to find out naming convention for ISDN4Linux channels!
>
> My home Asterisk is now functioning, but I still got some minor quirks to
> solve:
>
> 1. SIP softphone provides "Ringing" indication when I call another SIP
> softphone, but not when the call goes out via ISDN. All I get is "Trying"
> until called party picks up.
>
> 2. The |r option to Dial() always hungs up after half-a-second. This
happens
> with all types of channels, tried with Modem, SIP and IAX. I thought this
> option was meant to provide "Ringing" indication to caller?
>
> 3. Couldn't find out how to perform an action when extension goes
off-hook,
> before any digits are dialed. I'd like a specific extension to dial
> predefined number as soon as it goes off-hook (I'll have MGCP hardphone
for
> which I'll want to use this).
>
> 4. Tried calling US toll-free number using
> TRUNK=IAX2/gnophoneuser:gnophonepassword at iaxtel.com
> exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}@iaxtel)
> exten => _91800NXXXXXX,2,Congestion
>
> The call is connected but no audio comes thru - is the gateway still in
> operation btw?
>
>
> Any help greatly appreciated!
>
> TIA,
> Peter
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>