<p>Josh Soref has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/c/asterisk/+/17118">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">spelling: probably<br><br>Change-Id: I24c9fe9692db72f185ccb8306df0b4eab143cf37<br>---<br>M UPGRADE.txt<br>M channels/chan_sip.c<br>M include/asterisk/callerid.h<br>3 files changed, 4 insertions(+), 4 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/18/17118/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;"><span>diff --git a/UPGRADE.txt b/UPGRADE.txt</span><br><span>index 3ad163a..5cbbfee 100644</span><br><span>--- a/UPGRADE.txt</span><br><span>+++ b/UPGRADE.txt</span><br><span>@@ -2600,7 +2600,7 @@</span><br><span> </span><br><span> * The "canreinvite" option has changed. canreinvite=yes used to disable</span><br><span>   re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat</span><br><span style="color: hsl(0, 100%, 40%);">-  to disable re-invites when NAT=yes. This is propably what you want.</span><br><span style="color: hsl(120, 100%, 40%);">+  to disable re-invites when NAT=yes. This is probably what you want.</span><br><span>   The settings are now: "yes", "no", "nonat", "update". Please consult</span><br><span>   sip.conf.sample for detailed information.</span><br><span> </span><br><span>diff --git a/channels/chan_sip.c b/channels/chan_sip.c</span><br><span>index 654fdfd..eb9d4f1 100644</span><br><span>--- a/channels/chan_sip.c</span><br><span>+++ b/channels/chan_sip.c</span><br><span>@@ -114,7 +114,7 @@</span><br><span>  * \todo Add TCP/TLS information to function SIPPEER and CHANNEL function</span><br><span>  * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.</span><br><span>  *        The tcpbindaddr config option should only be used to open ADDITIONAL ports</span><br><span style="color: hsl(0, 100%, 40%);">- *    So we should propably go back to</span><br><span style="color: hsl(120, 100%, 40%);">+ *    So we should probably go back to</span><br><span>  *          bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP</span><br><span>  *                            if tlsenable=yes, open TLS port (provided we also have cert)</span><br><span>  *              tcpbindaddr = extra address for additional TCP connections</span><br><span>@@ -16452,7 +16452,7 @@</span><br><span>         p->refer->status = REFER_SENT;   /* Set refer status */</span><br><span> </span><br><span>    return transmit_invite(p, SIP_REFER, FALSE, 0, NULL);</span><br><span style="color: hsl(0, 100%, 40%);">-   /* We should propably wait for a NOTIFY here until we ack the transfer */</span><br><span style="color: hsl(120, 100%, 40%);">+     /* We should probably wait for a NOTIFY here until we ack the transfer */</span><br><span>    /* Maybe fork a new thread and wait for a STATUS of REFER_200OK on the refer status before returning to app_transfer */</span><br><span> </span><br><span>  /*! \todo In theory, we should hang around and wait for a reply, before</span><br><span>diff --git a/include/asterisk/callerid.h b/include/asterisk/callerid.h</span><br><span>index 777553d..7788aca 100644</span><br><span>--- a/include/asterisk/callerid.h</span><br><span>+++ b/include/asterisk/callerid.h</span><br><span>@@ -27,7 +27,7 @@</span><br><span> /*!</span><br><span>  * \page CID Caller ID names and numbers</span><br><span>  *</span><br><span style="color: hsl(0, 100%, 40%);">- * Caller ID names are currently 8 bit characters, propably</span><br><span style="color: hsl(120, 100%, 40%);">+ * Caller ID names are currently 8 bit characters, probably</span><br><span>  * ISO8859-1, depending on what your channel drivers handle.</span><br><span>  *</span><br><span>  * IAX2 and SIP caller ID names are UTF8</span><br><span></span><br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/c/asterisk/+/17118">change 17118</a>. To unsubscribe, or for help writing mail filters, 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/c/asterisk/+/17118"/><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-Change-Id: I24c9fe9692db72f185ccb8306df0b4eab143cf37 </div>
<div style="display:none"> Gerrit-Change-Number: 17118 </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>