]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
automerge commit
authorAutomerge script <automerge@asterisk.org>
Tue, 14 Feb 2006 01:03:47 +0000 (01:03 +0000)
committerAutomerge script <automerge@asterisk.org>
Tue, 14 Feb 2006 01:03:47 +0000 (01:03 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@9842 65c4cc65-6c06-0410-ace0-fbb531ad65f3

rtp.c

diff --git a/rtp.c b/rtp.c
index 2df27e287aabb6a0aa5a517e9d3d7cef0141537b..875d1be3f7a35da97cc1baf73297a34f438ff6a9 100644 (file)
--- a/rtp.c
+++ b/rtp.c
@@ -1,7 +1,7 @@
 /*
  * Asterisk -- An open source telephony toolkit.
  *
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Copyright (C) 1999 - 2006, Digium, Inc.
  *
  * Mark Spencer <markster@digium.com>
  *
@@ -811,17 +811,6 @@ int ast_rtp_lookup_code(struct ast_rtp* rtp, const int isAstFormat, const int co
                return rtp->rtp_lookup_code_cache_result;
        }
 
-       /* Check the dynamic list first */
-       for (pt = 0; pt < MAX_RTP_PT; ++pt) {
-               if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
-                       rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
-                       rtp->rtp_lookup_code_cache_code = code;
-                       rtp->rtp_lookup_code_cache_result = pt;
-                       return pt;
-               }
-       }
-
-       /* Then the static list */
        for (pt = 0; pt < MAX_RTP_PT; ++pt) {
                if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
                        rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;