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authormarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2005-02-03 03:30:41 +0000
committermarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2005-02-03 03:30:41 +0000
commitb52ae023fae000411d1ac87e4040bd7959ce85aa (patch)
tree1d1c6cc0c04d547a94ce2dad67156e84f9210784 /apps/app_dial.c
parent91a29e3b8abb81a1f669445bd211c4191d5850c3 (diff)
REname 'p' option to 'o' (bug #3490)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@4955 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'apps/app_dial.c')
-rwxr-xr-xapps/app_dial.c6
1 files changed, 3 insertions, 3 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c
index 98522abab..feb06db6e 100755
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -71,8 +71,8 @@ static char *descrip =
" making/redirecting the outgoing call. For example, some PSTNs\n"
" don't allow callerids from other extensions then the ones\n"
" that are assigned to you.\n"
-" 'p' -- Preserve the incoming Caller*ID on the outbound leg of the call (old style\n"
-" asterisk behavior)\n"
+" 'o' -- Original (inbound) Caller*ID should be placed on the outbound leg of the call
+" instead of using the destination extension (old style asterisk behavior)\n"
" 'r' -- indicate ringing to the calling party, pass no audio until answered.\n"
" 'm[(class)]' -- provide hold music to the calling party until answered (optionally\n"
" with the specified class.\n"
@@ -894,7 +894,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
ast_set2_flag(peerflags, strchr(transfer, 'W'), DIAL_MONITOR_OUT);
ast_set2_flag(peerflags, strchr(transfer, 'd'), DIAL_HALT_ON_DTMF);
ast_set2_flag(peerflags, strchr(transfer, 'g'), DIAL_GO_ON);
- ast_set2_flag(peerflags, strchr(transfer, 'p'), DIAL_PRESERVE_CALLERID);
+ ast_set2_flag(peerflags, strchr(transfer, 'o'), DIAL_PRESERVE_CALLERID);
}
strncpy(numsubst, number, sizeof(numsubst)-1);
/* If we're dialing by extension, look at the extension to know what to dial */