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authorKevin P. Fleming <kpfleming@digium.com>2006-06-01 12:43:01 +0000
committerKevin P. Fleming <kpfleming@digium.com>2006-06-01 12:43:01 +0000
commit6bce269454aa4ed1b20da0031f5823e84d2db1d2 (patch)
tree861d557ba45039bf7c163abb13133cb6613b67ea /configs/sip.conf.sample
parent073e346f4e1ca7e2f360de24f55d2db1a3e0acda (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: https://origsvn.digium.com/svn/asterisk/trunk@31322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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