<p>Jenkins2 <strong>merged</strong> this change.</p><p><a href="https://gerrit.asterisk.org/8157">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 bf859fe..310ff20 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>@@ -1111,7 +1111,7 @@<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>@@ -1316,7 +1316,7 @@<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>@@ -1361,7 +1361,7 @@<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>@@ -1445,8 +1445,8 @@<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>@@ -1486,7 +1486,7 @@<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>@@ -1560,7 +1560,7 @@<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>@@ -1659,7 +1659,7 @@<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>@@ -1785,7 +1785,7 @@<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/8157">change 8157</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/8157"/><meta itemprop="name" content="View Change"/></div></div>

<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: master </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: 8157 </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>