https://origsvn.digium.com/svn/asterisk/branches/1.2
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r31321 | kpfleming | 2006-06-01 07:41:47 -0500 (Thu, 01 Jun 2006) | 2 lines
remove a sample entry that never should have been added (code to support it was not merged)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31322
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
-;astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists
;callingpres=allowed_passed_screen ; Set caller ID presentation
; See README.callingpres for more information