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authorJoshua Colp <jcolp@digium.com>2016-11-28 19:00:32 -0600
committerGerrit Code Review <gerrit2@gerrit.digium.api>2016-11-28 19:00:32 -0600
commitfdf4355bd054ce3a264f3d7a83725259b8acb32f (patch)
tree06f877791d0c038c61e372fe83a1b6ba8351364c
parent350a9a6bd4cadc97700c7dd97c5c603207fe0cf1 (diff)
parenta3f48be0da4f1261cfcfd548aaf2c8778ee6da43 (diff)
Merge "res/res_pjsip: Fix documentation whitespace issues"
-rw-r--r--res/res_pjsip.c16
1 files changed, 8 insertions, 8 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 98530cb7b..8c2d37137 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -925,14 +925,14 @@
On outbound requests, force the user portion of the Contact header to this value.
</para></description>
</configOption>
- <configOption name="asymmetric_rtp_codec" default="no">
- <synopsis>Allow the sending and receiving RTP codec to differ</synopsis>
- <description><para>
- When set to "yes" the codec in use for sending will be allowed to differ from
- that of the received one. PJSIP will not automatically switch the sending one
- to the receiving one.
- </para></description>
- </configOption>
+ <configOption name="asymmetric_rtp_codec" default="no">
+ <synopsis>Allow the sending and receiving RTP codec to differ</synopsis>
+ <description><para>
+ When set to "yes" the codec in use for sending will be allowed to differ from
+ that of the received one. PJSIP will not automatically switch the sending one
+ to the receiving one.
+ </para></description>
+ </configOption>
</configObject>
<configObject name="auth">
<synopsis>Authentication type</synopsis>