Merged revisions 52645 via svnmerge from
authorRussell Bryant <russell@russellbryant.com>
Mon, 29 Jan 2007 21:27:34 +0000 (21:27 +0000)
committerRussell Bryant <russell@russellbryant.com>
Mon, 29 Jan 2007 21:27:34 +0000 (21:27 +0000)
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) | 6 lines

Fix a problem with packet-to-packet bridging and DTMF mode translation.  P2P
bridging can only be used when the DTMF modes don't match if the core is
monitoring DTMF in both directions.  Then, the core will handle the translation.
Otherwise, this bridging method can not be used.
(issue #8936)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52646 65c4cc65-6c06-0410-ace0-fbb531ad65f3

main/rtp.c

index ba5e11d..709fef5 100644 (file)
@@ -3308,11 +3308,19 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
         * |-----------|------------|-----------------------|
         * | Inband    | False      | True                  |
         * | RFC2833   | True       | True                  |
-        * | SIP Info  | False      | False                 |
+        * | SIP INFO  | False      | False                 |
         * --------------------------------------------------
+        * However, if DTMF from both channels is being monitored by the core, then
+        * we can still do packet-to-packet bridging, because passing through the 
+        * core will handle DTMF mode translation.
         */
        if ( (ast_test_flag(p0, FLAG_HAS_DTMF) != ast_test_flag(p1, FLAG_HAS_DTMF)) ||
                 (!c0->tech->send_digit_begin != !c1->tech->send_digit_begin)) {
+               if (!ast_test_flag(p0, FLAG_P2P_NEED_DTMF) || !ast_test_flag(p1, FLAG_P2P_NEED_DTMF)) {
+                       ast_channel_unlock(c0);
+                       ast_channel_unlock(c1);
+                       return AST_BRIDGE_FAILED_NOWARN;
+               }
                audio_p0_res = AST_RTP_TRY_PARTIAL;
                audio_p1_res = AST_RTP_TRY_PARTIAL;
        }