<p>Jenkins2 <strong>merged</strong> this change.</p><p><a href="https://gerrit.asterisk.org/8155">View Change</a></p><div style="white-space:pre-wrap">Approvals:
Corey Farrell: Looks good to me, but someone else must approve
Sean Bright: Looks good to me, approved
Jenkins2: Approved for Submit
</div><pre style="font-family: monospace,monospace; white-space: pre-wrap;">res_pjsip.c: Fix documentation typos.<br><br>Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068<br>---<br>M res/res_pjsip.c<br>1 file changed, 14 insertions(+), 14 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">diff --git a/res/res_pjsip.c b/res/res_pjsip.c<br>index 8ec89c6..7c8aabf 100644<br>--- a/res/res_pjsip.c<br>+++ b/res/res_pjsip.c<br>@@ -66,7 +66,7 @@<br> It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis><br> dialable entries of their own. Communication with another SIP device is<br> accomplished via Addresses of Record (AoRs) which have one or more<br>- contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to<br>+ contacts associated with them. Endpoints <emphasis>NOT</emphasis> configured to<br> use a <literal>transport</literal> will default to first transport found<br> in <filename>pjsip.conf</filename> that matches its type.<br> </para><br>@@ -449,7 +449,7 @@<br> <configOption name="timers_min_se" default="90"><br> <synopsis>Minimum session timers expiration period</synopsis><br> <description><para><br>- Minimium session timer expiration period. Time in seconds.<br>+ Minimum session timer expiration period. Time in seconds.<br> </para></description><br> </configOption><br> <configOption name="timers" default="yes"><br>@@ -467,7 +467,7 @@<br> <configOption name="timers_sess_expires" default="1800"><br> <synopsis>Maximum session timer expiration period</synopsis><br> <description><para><br>- Maximium session timer expiration period. Time in seconds.<br>+ Maximum session timer expiration period. Time in seconds.<br> </para></description><br> </configOption><br> <configOption name="transport"><br>@@ -509,7 +509,7 @@<br> <synopsis>Must be of type 'endpoint'.</synopsis><br> </configOption><br> <configOption name="use_ptime" default="no"><br>- <synopsis>Use Endpoint's requested packetisation interval</synopsis><br>+ <synopsis>Use Endpoint's requested packetization interval</synopsis><br> </configOption><br> <configOption name="use_avpf" default="no"><br> <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this<br>@@ -647,7 +647,7 @@<br> Forward error correction should be used.<br> </para></enum><br> <enum name="redundancy"><para><br>- Redundacy error correction should be used.<br>+ Redundancy error correction should be used.<br> </para></enum><br> </enumlist><br> </description><br>@@ -1058,7 +1058,7 @@<br> <description><para>Only used when auth_type is <literal>md5</literal>.</para></description><br> </configOption><br> <configOption name="password"><br>- <synopsis>PlainText password used for authentication.</synopsis><br>+ <synopsis>Plain text password used for authentication.</synopsis><br> <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description><br> </configOption><br> <configOption name="realm"><br>@@ -1263,7 +1263,7 @@<br> </description><br> </configOption><br> <configOption name="symmetric_transport" default="no"><br>- <synopsis>Use the same transport for outgoing reqests as incoming ones.</synopsis><br>+ <synopsis>Use the same transport for outgoing requests as incoming ones.</synopsis><br> <description><br> <para>When a request from a dynamic contact<br> comes in on a transport with this option set to 'yes',<br>@@ -1308,7 +1308,7 @@<br> <configOption name="qualify_timeout" default="3.0"><br> <synopsis>Timeout for qualify</synopsis><br> <description><para><br>- If the contact doesn't repond to the OPTIONS request before the timeout,<br>+ If the contact doesn't respond to the OPTIONS request before the timeout,<br> the contact is marked unavailable.<br> If <literal>0</literal> no timeout. Time in fractional seconds.<br> </para></description><br>@@ -1392,8 +1392,8 @@<br> <literal>endpoint</literal> for calls.<br> </para><para><br> This can be used as another way of grouping a list of contacts to dial<br>- rather than specifing them each directly when dialing via the dialplan.<br>- This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.<br>+ rather than specifying them each directly when dialing via the dialplan.<br>+ This must be used in conjunction with the <literal>PJSIP_DIAL_CONTACTS</literal>.<br> </para><para><br> Registrations: For Asterisk to match an inbound registration to an endpoint,<br> the AoR object name must match the user portion of the SIP URI in the "To:"<br>@@ -1433,7 +1433,7 @@<br> <configOption name="maximum_expiration" default="7200"><br> <synopsis>Maximum time to keep an AoR</synopsis><br> <description><para><br>- Maximium time to keep a peer with explicit expiration. Time in seconds.<br>+ Maximum time to keep a peer with explicit expiration. Time in seconds.<br> </para></description><br> </configOption><br> <configOption name="max_contacts" default="0"><br>@@ -1507,7 +1507,7 @@<br> <configOption name="qualify_timeout" default="3.0"><br> <synopsis>Timeout for qualify</synopsis><br> <description><para><br>- If the contact doesn't repond to the OPTIONS request before the timeout,<br>+ If the contact doesn't respond to the OPTIONS request before the timeout,<br> the contact is marked unavailable.<br> If <literal>0</literal> no timeout. Time in fractional seconds.<br> </para></description><br>@@ -1606,7 +1606,7 @@<br> <configOption name="disable_multi_domain" default="no"><br> <synopsis>Disable Multi Domain support</synopsis><br> <description><para><br>- If disabled it can improve realtime performace by reducing number of database requsts.<br>+ If disabled it can improve realtime performance by reducing the number of database requests.<br> </para></description><br> </configOption><br> <configOption name="max_initial_qualify_time" default="0"><br>@@ -1732,7 +1732,7 @@<br> in the user field of a SIP URI then the field is truncated<br> at the first semicolon. This effectively makes the semicolon<br> a non-usable character for PJSIP endpoint names, extensions,<br>- and AORs. This can be useful for improving compatability with<br>+ and AORs. This can be useful for improving compatibility with<br> an ITSP that likes to use user options for whatever reason.<br> </para><br> <example title="Sample SIP URI"><br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/8155">change 8155</a>. To unsubscribe, visit <a href="https://gerrit.asterisk.org/settings">settings</a>.</p><div itemscope itemtype="http://schema.org/EmailMessage"><div itemscope itemprop="action" itemtype="http://schema.org/ViewAction"><link itemprop="url" href="https://gerrit.asterisk.org/8155"/><meta itemprop="name" content="View Change"/></div></div>
<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: 13 </div>
<div style="display:none"> Gerrit-MessageType: merged </div>
<div style="display:none"> Gerrit-Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068 </div>
<div style="display:none"> Gerrit-Change-Number: 8155 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Richard Mudgett <rmudgett@digium.com> </div>
<div style="display:none"> Gerrit-Reviewer: Corey Farrell <git@cfware.com> </div>
<div style="display:none"> Gerrit-Reviewer: Jenkins2 </div>
<div style="display:none"> Gerrit-Reviewer: Sean Bright <sean.bright@gmail.com> </div>