All,
I'm trying to hear the asterisk's auto attendant in its default
configuration. According to VoIP Hacks in Chapter 4, I found the
following excerpt after successfully configuring my SIP IP Phone (Cisco
7960G):
In its default configuration, Asterisk has an auto-attendant that can
route calls. To try it out, take the IP phone off the hook and dial 2.
Then dial the BudgeTone's Send button. You will hear a friendly voice
saying, "Asterisk is an open source, fully featured PBX and IVR
platform?."
However, when I dial '2' on the phone, I just get a busy signal. Through
the CLI it looks to have the demo available:
vitamin-nybw*CLI> console dial 2
[Apr 24 12:34:35] WARNING[8070]: chan_oss.c:682 setformat: Unable to
re-open DSP device /dev/dsp: No such file or directory
-- Executing [2@default:1] BackGround("OSS/dsp",
"demo-moreinfo") in
new stack
<< Console call has been answered >>
-- <OSS/dsp> Playing 'demo-moreinfo' (language 'en')
[Apr 24 12:34:36] WARNING[8071]: chan_oss.c:682 setformat: Unable to
re-open DSP device /dev/dsp: No such file or directory
Any idea why I can't hear the asterisk default demo when dialing 2?
- sf