[Asterisk-Users] new asterisk server...welcome message cut off

[Rod Morison]

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 at default:1] BackGround("SIP/159-f2da", 
"demo-moreinfo") in new stack
    -- Playing 'demo-moreinfo' (language 'en')
    -- Executing [2 at default:2] Goto("SIP/159-f2da", "s|instruct") in new 
stack
    -- Goto (default,s,6)
    -- Executing [s at default:6] BackGround("SIP/159-f2da", 
"demo-instruct") in new stack
    -- Playing 'demo-instruct' (language 'en')
    -- Executing [s at default:7] WaitExten("SIP/159-f2da", "") in new stack
    -- Timeout on SIP/159-f2da, going to 't'
    -- Executing [t at 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'