Merged revisions 31321 via svnmerge from
authorKevin P. Fleming <kpfleming@digium.com>
Thu, 1 Jun 2006 12:43:01 +0000 (12:43 +0000)
committerKevin P. Fleming <kpfleming@digium.com>
Thu, 1 Jun 2006 12:43:01 +0000 (12:43 +0000)
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

configs/sip.conf.sample

index 8a52b07..67d6dd2 100644 (file)
@@ -460,7 +460,6 @@ srvlookup=yes                       ; Enable DNS SRV lookups on outbound calls
 ;allow=alaw
 ;allow=g723.1                  ; Asterisk only supports g723.1 pass-thru!
 ;allow=g729                    ; Pass-thru only unless g729 license obtained
 ;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
 
 ;callingpres=allowed_passed_screen     ; Set caller ID presentation
                                ; See README.callingpres for more information