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