asterisk users - Jun 2006 - new asterisk server...welcome message cut off

I just brought up an asterisk server. On dialing "2" from grandstream 
hardphone, I get the beginning of the welcome message, but each segment 
is cutoff. Specifically

"Asterisk is an open source full"-1s silence-"if you'd like
to learn
more technical information about Asterisk"-11s silience-"goodbye"

Any help or pointers on how to gather more debug info is appreciated in 
advance! Here's the output from -vvvc for the call:


    -- Executing [2@default:1] BackGround("SIP/159-f2da", 
"demo-moreinfo") in new stack
    -- Playing 'demo-moreinfo' (language 'en')
    -- Executing [2@default:2] Goto("SIP/159-f2da",
"s|instruct") in new
stack
    -- Goto (default,s,6)
    -- Executing [s@default:6] BackGround("SIP/159-f2da", 
"demo-instruct") in new stack
    -- Playing 'demo-instruct' (language 'en')
    -- Executing [s@default:7] WaitExten("SIP/159-f2da", "")
in new stack
    -- Timeout on SIP/159-f2da, going to 't'
    -- Executing [t@default:1] Goto("SIP/159-f2da", "#|1")
in new stack
    -- Goto (default,#,1)
    -- Executing [#@default:1] Playback("SIP/159-f2da",
"demo-thanks")
in new stack
    -- Playing 'demo-thanks' (language 'en')
    -- Executing [#@default:2] Hangup("SIP/159-f2da", "") in
new stack
  == Spawn extension (default, #, 2) exited non-zero on 'SIP/159-f2da'

Same behavior from xten. I downloaded/built the 1.2 branch and don't see 
the same problem (I'm running on ppc linux, certainly a minority config).

I know it's on the wiki somewhere, but how do I submit a bug?

Thanks,

Rod

Brian Vincent (C) wrote:
> Have you tried using another client to see if it does the same thing?  I
> like using the Xten soft phone to test with.  It's simple to configure
> and works well - www.xten.com
>
> -------------------
> Brian Vincent
> Copper Mountain Telecom
> vincentb@coppercolorado.com
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rod
> Morison
> Sent: Wednesday, June 21, 2006 6:29 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] new asterisk server...welcome message cut off
>
> I just brought up an asterisk server. On dialing "2" from
grandstream
> hardphone, I get the beginning of the welcome message, but each segment 
> is cutoff. Specifically
>
> "Asterisk is an open source full"-1s silence-"if you'd
like to learn
> more technical information about Asterisk"-11s
silience-"goodbye"
>
>