[Asterisk-code-review] res/res pjsip: Fix documentation whitespace issues (asterisk[13])
Matt Jordan
asteriskteam at digium.com
Mon Nov 28 15:12:57 CST 2016
Matt Jordan has uploaded a new change for review. ( https://gerrit.asterisk.org/4512 )
Change subject: res/res_pjsip: Fix documentation whitespace issues
......................................................................
res/res_pjsip: Fix documentation whitespace issues
Tabs > Spaces.
Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0
---
M res/res_pjsip.c
1 file changed, 8 insertions(+), 8 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/12/4512/1
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index dd4a619..d83bd06 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -919,14 +919,14 @@
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>
--
To view, visit https://gerrit.asterisk.org/4512
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Gerrit-MessageType: newchange
Gerrit-Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Matt Jordan <mjordan at digium.com>
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