diff options
| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-18 17:13:32 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-18 17:13:32 +0000 |
| commit | d6da2dcdb870a31b9ad6edd98538814f482d3662 (patch) | |
| tree | 9bf951bc5f22140e47c68e5b110ab39e1623a61f /main/rtp.c | |
| parent | 17d5723603d66983f3e3c74bee98e69ec72c2c63 (diff) | |
Merged revisions 165591 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines
Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
(closes issue #13545)
Reported by: davidw
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165599 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'main/rtp.c')
| -rw-r--r-- | main/rtp.c | 5 |
1 files changed, 2 insertions, 3 deletions
diff --git a/main/rtp.c b/main/rtp.c index c613aac17..904944359 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -2072,10 +2072,9 @@ int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1) else destcodec = 0; /* Ensure we have at least one matching codec */ - if (!(srccodec & destcodec)) { + if (srcp && !(srccodec & destcodec)) { ast_channel_unlock(c0); - if (c1) - ast_channel_unlock(c1); + ast_channel_unlock(c1); return 0; } /* Consider empty media as non-existent */ |