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List:       asterisk-commits
Subject:    [asterisk-commits] =?utf-8?q?res/res_pjsip=3A_Fix_documentation_w?= =?utf-8?q?hitespace_issues_=28as
From:       SVN commits to the Asterisk project <asterisk-commits () lists ! digium ! com>
Date:       2016-11-29 1:25:15
Message-ID: mailman.37871.1480451398.1232.asterisk-commits () lists ! digium ! com
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Joshua Colp has submitted this change and it was merged. ( \
https://gerrit.asterisk.org/4513 )

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(-)

Approvals:
  Richard Mudgett: Looks good to me, but someone else must approve
  Anonymous Coward #1000019: Verified
  Joshua Colp: Looks good to me, approved



diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index ea81cd5..b1114a4 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -922,14 +922,14 @@
 						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/4513
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: merged
Gerrit-Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Matt Jordan <mjordan@digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp@digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett@digium.com>

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