<p>Josh Soref has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/c/asterisk/+/16759">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">spelling: available<br><br>Change-Id: I320978b3ca2e72e3d0fb469b0a68c7651e96e8de<br>---<br>M CHANGES<br>M apps/app_jack.c<br>M apps/app_queue.c<br>M cdr/cdr_beanstalkd.c<br>M cel/cel_beanstalkd.c<br>M channels/chan_dahdi.c<br>M main/pbx.c<br>7 files changed, 7 insertions(+), 7 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/59/16759/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;"><span>diff --git a/CHANGES b/CHANGES</span><br><span>index e4400c2..05ec24e 100644</span><br><span>--- a/CHANGES</span><br><span>+++ b/CHANGES</span><br><span>@@ -926,7 +926,7 @@</span><br><span> </span><br><span> res_rtp_asterisk</span><br><span> ------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * The existing strictrtp option in rtp.conf has a new choice availabe, called</span><br><span style="color: hsl(120, 100%, 40%);">+ * The existing strictrtp option in rtp.conf has a new choice available, called</span><br><span> 'seqno', which behaves the same way as setting strictrtp to 'yes', but will</span><br><span> ignore the time interval during learning so that bursts of packets can still</span><br><span> trigger learning our source.</span><br><span>diff --git a/apps/app_jack.c b/apps/app_jack.c</span><br><span>index 1609972..1abe490 100644</span><br><span>--- a/apps/app_jack.c</span><br><span>+++ b/apps/app_jack.c</span><br><span>@@ -607,7 +607,7 @@</span><br><span> * \param[in] jack_data This is the jack_data struct that contains the input</span><br><span> * ringbuffer that audio will be read from.</span><br><span> * \param[out] out_frame If this argument is non-NULL, then assuming there is</span><br><span style="color: hsl(0, 100%, 40%);">- * enough data avilable in the ringbuffer, the audio in this frame</span><br><span style="color: hsl(120, 100%, 40%);">+ * enough data available in the ringbuffer, the audio in this frame</span><br><span> * will get replaced with audio from the input buffer. If there is</span><br><span> * not enough data available to read at this time, then the frame</span><br><span> * data gets zeroed out.</span><br><span>diff --git a/apps/app_queue.c b/apps/app_queue.c</span><br><span>index b0f463b..b15d3cf 100644</span><br><span>--- a/apps/app_queue.c</span><br><span>+++ b/apps/app_queue.c</span><br><span>@@ -5576,7 +5576,7 @@</span><br><span> res = 0;</span><br><span> }</span><br><span> </span><br><span style="color: hsl(0, 100%, 40%);">- /* Update realtime members if this is the first call and number of avalable members is 0 */</span><br><span style="color: hsl(120, 100%, 40%);">+ /* Update realtime members if this is the first call and number of available members is 0 */</span><br><span> if (avl == 0 && qe->pos == 1) {</span><br><span> update_realtime_members(qe->parent);</span><br><span> }</span><br><span>diff --git a/cdr/cdr_beanstalkd.c b/cdr/cdr_beanstalkd.c</span><br><span>index 524274f..107cdfc 100644</span><br><span>--- a/cdr/cdr_beanstalkd.c</span><br><span>+++ b/cdr/cdr_beanstalkd.c</span><br><span>@@ -20,7 +20,7 @@</span><br><span> * \file</span><br><span> * \brief Asterisk Beanstalkd CDR records.</span><br><span> *</span><br><span style="color: hsl(0, 100%, 40%);">- * This module requires the beanstalk-client library, avaialble from</span><br><span style="color: hsl(120, 100%, 40%);">+ * This module requires the beanstalk-client library, available from</span><br><span> * https://github.com/deepfryed/beanstalk-client</span><br><span> *</span><br><span> * See also</span><br><span>diff --git a/cel/cel_beanstalkd.c b/cel/cel_beanstalkd.c</span><br><span>index fe4f430..1bfe2e8 100644</span><br><span>--- a/cel/cel_beanstalkd.c</span><br><span>+++ b/cel/cel_beanstalkd.c</span><br><span>@@ -22,7 +22,7 @@</span><br><span> *</span><br><span> * \brief Asterisk Channel Event Beanstalkd backend</span><br><span> *</span><br><span style="color: hsl(0, 100%, 40%);">- * This module requires the beanstalk-client library, avaialble from</span><br><span style="color: hsl(120, 100%, 40%);">+ * This module requires the beanstalk-client library, available from</span><br><span> * https://github.com/deepfryed/beanstalk-client</span><br><span> *</span><br><span> * See also</span><br><span>diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c</span><br><span>index 74c76f7..e750648 100644</span><br><span>--- a/channels/chan_dahdi.c</span><br><span>+++ b/channels/chan_dahdi.c</span><br><span>@@ -6234,7 +6234,7 @@</span><br><span> p->subs[SUB_REAL].inthreeway = 0;</span><br><span> }</span><br><span> } else if (idx == SUB_CALLWAIT) {</span><br><span style="color: hsl(0, 100%, 40%);">- /* Ditch the holding callwait call, and immediately make it availabe */</span><br><span style="color: hsl(120, 100%, 40%);">+ /* Ditch the holding callwait call, and immediately make it available */</span><br><span> if (p->subs[SUB_CALLWAIT].inthreeway) {</span><br><span> /* This is actually part of a three way, placed on hold. Place the third part</span><br><span> on music on hold now */</span><br><span>diff --git a/main/pbx.c b/main/pbx.c</span><br><span>index d0ee127..dbab50e 100644</span><br><span>--- a/main/pbx.c</span><br><span>+++ b/main/pbx.c</span><br><span>@@ -4330,7 +4330,7 @@</span><br><span> callid = ast_read_threadstorage_callid();</span><br><span> /* If the thread isn't already associated with a callid, we should create that association. */</span><br><span> if (!callid) {</span><br><span style="color: hsl(0, 100%, 40%);">- /* Associate new PBX thread with the channel call id if it is availble.</span><br><span style="color: hsl(120, 100%, 40%);">+ /* Associate new PBX thread with the channel call id if it is available.</span><br><span> * If not, create a new one instead.</span><br><span> */</span><br><span> callid = ast_channel_callid(c);</span><br><span></span><br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/c/asterisk/+/16759">change 16759</a>. 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<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: master </div>
<div style="display:none"> Gerrit-Change-Id: I320978b3ca2e72e3d0fb469b0a68c7651e96e8de </div>
<div style="display:none"> Gerrit-Change-Number: 16759 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Josh Soref <jsoref@gmail.com> </div>
<div style="display:none"> Gerrit-CC: Friendly Automation </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>