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authorkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2006-06-01 12:43:01 +0000
committerkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2006-06-01 12:43:01 +0000
commitd42fc385b543fc9b71eb971c77ea822a88b22b30 (patch)
tree861d557ba45039bf7c163abb13133cb6613b67ea /configs/sip.conf.sample
parenta886139b21dba0216790453a13bd816228bf13ad (diff)
Merged revisions 31321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31322 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/sip.conf.sample')
-rw-r--r--configs/sip.conf.sample1
1 files changed, 0 insertions, 1 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 8a52b07a8..67d6dd21f 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -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
-;astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists
;callingpres=allowed_passed_screen ; Set caller ID presentation
; See README.callingpres for more information