From 6bce269454aa4ed1b20da0031f5823e84d2db1d2 Mon Sep 17 00:00:00 2001 From: "Kevin P. Fleming" Date: Thu, 1 Jun 2006 12:43:01 +0000 Subject: 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 --- configs/sip.conf.sample | 1 - 1 file changed, 1 deletion(-) (limited to 'configs/sip.conf.sample') 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 -- cgit v1.2.3