]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Add missing carriage return and line feed to the SDP line indicating that we don...
authorRussell Bryant <russell@russellbryant.com>
Thu, 17 Nov 2005 20:25:40 +0000 (20:25 +0000)
committerRussell Bryant <russell@russellbryant.com>
Thu, 17 Nov 2005 20:25:40 +0000 (20:25 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7130 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ChangeLog
channels/chan_sip.c

index 37caadb193df259b21f68224f1d1bfafee9fb65f..621cbd2b81c2a47ad32d1129d3e2bec912076adc 100755 (executable)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,7 @@
+2005-11-16  Russell Bryant  <russell@digium.com>
+
+       * channels/chan_sip.c: Add missing carriage return and line feed to the SDP line indicating that we don't support VAD (issue #5780)
+
 2005-11-16  Kevin P. Fleming  <kpfleming@digium.com>
 
        * Asterisk 1.2.0 released.
index d44655a953b5f3ad4e287d1141c87b4871fd676f..c3b6e9b1dab47ae6bcf600a225a2baf186c08fab 100755 (executable)
@@ -4267,7 +4267,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
                         sample_rate);
        if (codec == AST_FORMAT_G729A)
                /* Indicate that we don't support VAD (G.729 annex B) */
-               ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no", rtp_code);
+               ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code);
 }
 
 static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,