From 66d2f2de66c12bd00d5d039cfe0becb3289d6acb Mon Sep 17 00:00:00 2001
From: Martin Pycko <martinp@digium.com>
Date: Tue, 25 Nov 2003 16:26:15 +0000
Subject: [PATCH] Warn about not being able to do reinvite in the right place
and unlock the mutexes before returning
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
rtp.c | 7 +++++--
1 file changed, 5 insertions(+), 2 deletions(-)
diff --git a/rtp.c b/rtp.c
index 82a1cd5f6e..92fddcafe1 100755
--- a/rtp.c
+++ b/rtp.c
@@ -1199,10 +1199,13 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
int codec0,codec1;
codec0 = pr0->get_codec(c0);
codec1 = pr1->get_codec(c1);
- ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, can't do reinvite\n",codec0,codec1);
/* Hey, we can't do reinvite if both parties speak diffrent codecs */
- if (codec0 != codec1)
+ if (codec0 != codec1) {
+ ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, can't do reinvite\n",codec0,codec1);
+ ast_mutex_unlock(&c0->lock);
+ ast_mutex_unlock(&c1->lock);
return -2;
+ }
}
if (pr0->set_rtp_peer(c0, p1, vp1))
ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
--
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