[Asterisk-Users] Strange console call problem

[Scott Williamson]

Hi,

When I dial from my sip device to the extension 1234 which is linked to
the ALSA console driver the call fails with the message "No channel type
registered for 'ALSA'" (see below).

I would like to have the console autoanswer for paging. 

However when I call from the console to the sip device the call
completes fine.

I alias alsa device hw:1,0 to card1 in /etc/asound.conf to get rid of
the pesky ',' (see below)

Any ideas on why this would happen?

I am using Asterisk 1.0.5 on Debian testing (package version 1:1.0.5-2,
which is the latest)

--snip SIP to Console--

    -- Executing Macro("SIP/2201-621f", "stdexten|1234|ALSA/card1") in
new stack
    -- Executing Dial("SIP/2201-621f", "ALSA/card1|20") in new stack
Mar 17 12:24:33 WARNING[23967]: channel.c:1901 ast_request: No channel
type registered for 'ALSA'
Mar 17 12:24:33 NOTICE[23967]: app_dial.c:749 dial_exec: Unable to
create channel of type 'ALSA'

--snip--

--snip Console to SIP--

dial 2201
    -- Executing Macro("ALSA/card1", "stdexten|2201|SIP/2201") in new
stack
    -- Executing Dial("ALSA/card1", "SIP/2201|20") in new stack
    -- Called 2201
Urgent handler
Urgent handler
    -- SIP/2201-506f is ringing
Urgent handler
    -- SIP/2201-506f answered ALSA/card1
 << Console call has been answered >>
Urgent handler
 << Hangup on console >>

--snip--

--asound.conf--
pcm.card1 {
        type hw
        card 1
        }
--end-- 

--alsa.conf--
...
; To set which ALSA device to use, change this parameter
input_device=card1
output_device=card1
...
--end--


--
Scott J. Williamson <scott at sjwilliamson.ca>

I wouldn't ever write the full sentence myself, but then, I never use
goto either. -- Larry Wall in <199709032332.QAA21669 at wall.org>