[Asterisk-code-review] res pjsip.c: Fix documentation typos. (asterisk[15])
Jenkins2
asteriskteam at digium.com
Sun Feb 4 12:08:07 CST 2018
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/8156 )
Change subject: res_pjsip.c: Fix documentation typos.
......................................................................
res_pjsip.c: Fix documentation typos.
Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068
---
M res/res_pjsip.c
1 file changed, 14 insertions(+), 14 deletions(-)
Approvals:
Corey Farrell: Looks good to me, but someone else must approve
Sean Bright: Looks good to me, approved
Jenkins2: Approved for Submit
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index de0afc4..7867c52 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -66,7 +66,7 @@
It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis>
dialable entries of their own. Communication with another SIP device is
accomplished via Addresses of Record (AoRs) which have one or more
- contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
+ contacts associated with them. Endpoints <emphasis>NOT</emphasis> configured to
use a <literal>transport</literal> will default to first transport found
in <filename>pjsip.conf</filename> that matches its type.
</para>
@@ -449,7 +449,7 @@
<configOption name="timers_min_se" default="90">
<synopsis>Minimum session timers expiration period</synopsis>
<description><para>
- Minimium session timer expiration period. Time in seconds.
+ Minimum session timer expiration period. Time in seconds.
</para></description>
</configOption>
<configOption name="timers" default="yes">
@@ -467,7 +467,7 @@
<configOption name="timers_sess_expires" default="1800">
<synopsis>Maximum session timer expiration period</synopsis>
<description><para>
- Maximium session timer expiration period. Time in seconds.
+ Maximum session timer expiration period. Time in seconds.
</para></description>
</configOption>
<configOption name="transport">
@@ -509,7 +509,7 @@
<synopsis>Must be of type 'endpoint'.</synopsis>
</configOption>
<configOption name="use_ptime" default="no">
- <synopsis>Use Endpoint's requested packetisation interval</synopsis>
+ <synopsis>Use Endpoint's requested packetization interval</synopsis>
</configOption>
<configOption name="use_avpf" default="no">
<synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
@@ -647,7 +647,7 @@
Forward error correction should be used.
</para></enum>
<enum name="redundancy"><para>
- Redundacy error correction should be used.
+ Redundancy error correction should be used.
</para></enum>
</enumlist>
</description>
@@ -1111,7 +1111,7 @@
<description><para>Only used when auth_type is <literal>md5</literal>.</para></description>
</configOption>
<configOption name="password">
- <synopsis>PlainText password used for authentication.</synopsis>
+ <synopsis>Plain text password used for authentication.</synopsis>
<description><para>Only used when auth_type is <literal>userpass</literal>.</para></description>
</configOption>
<configOption name="realm">
@@ -1316,7 +1316,7 @@
</description>
</configOption>
<configOption name="symmetric_transport" default="no">
- <synopsis>Use the same transport for outgoing reqests as incoming ones.</synopsis>
+ <synopsis>Use the same transport for outgoing requests as incoming ones.</synopsis>
<description>
<para>When a request from a dynamic contact
comes in on a transport with this option set to 'yes',
@@ -1361,7 +1361,7 @@
<configOption name="qualify_timeout" default="3.0">
<synopsis>Timeout for qualify</synopsis>
<description><para>
- If the contact doesn't repond to the OPTIONS request before the timeout,
+ If the contact doesn't respond to the OPTIONS request before the timeout,
the contact is marked unavailable.
If <literal>0</literal> no timeout. Time in fractional seconds.
</para></description>
@@ -1445,8 +1445,8 @@
<literal>endpoint</literal> for calls.
</para><para>
This can be used as another way of grouping a list of contacts to dial
- rather than specifing them each directly when dialing via the dialplan.
- This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
+ rather than specifying them each directly when dialing via the dialplan.
+ This must be used in conjunction with the <literal>PJSIP_DIAL_CONTACTS</literal>.
</para><para>
Registrations: For Asterisk to match an inbound registration to an endpoint,
the AoR object name must match the user portion of the SIP URI in the "To:"
@@ -1486,7 +1486,7 @@
<configOption name="maximum_expiration" default="7200">
<synopsis>Maximum time to keep an AoR</synopsis>
<description><para>
- Maximium time to keep a peer with explicit expiration. Time in seconds.
+ Maximum time to keep a peer with explicit expiration. Time in seconds.
</para></description>
</configOption>
<configOption name="max_contacts" default="0">
@@ -1560,7 +1560,7 @@
<configOption name="qualify_timeout" default="3.0">
<synopsis>Timeout for qualify</synopsis>
<description><para>
- If the contact doesn't repond to the OPTIONS request before the timeout,
+ If the contact doesn't respond to the OPTIONS request before the timeout,
the contact is marked unavailable.
If <literal>0</literal> no timeout. Time in fractional seconds.
</para></description>
@@ -1659,7 +1659,7 @@
<configOption name="disable_multi_domain" default="no">
<synopsis>Disable Multi Domain support</synopsis>
<description><para>
- If disabled it can improve realtime performace by reducing number of database requsts.
+ If disabled it can improve realtime performance by reducing the number of database requests.
</para></description>
</configOption>
<configOption name="max_initial_qualify_time" default="0">
@@ -1785,7 +1785,7 @@
in the user field of a SIP URI then the field is truncated
at the first semicolon. This effectively makes the semicolon
a non-usable character for PJSIP endpoint names, extensions,
- and AORs. This can be useful for improving compatability with
+ and AORs. This can be useful for improving compatibility with
an ITSP that likes to use user options for whatever reason.
</para>
<example title="Sample SIP URI">
--
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Gerrit-Project: asterisk
Gerrit-Branch: 15
Gerrit-MessageType: merged
Gerrit-Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068
Gerrit-Change-Number: 8156
Gerrit-PatchSet: 1
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Corey Farrell <git at cfware.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Sean Bright <sean.bright at gmail.com>
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