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res_pjsip.c: Fix documentation typos.

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Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068
This commit is contained in:
Richard Mudgett
2018-02-02 15:11:27 -06:00
parent 4142eacfef
commit 0960de71ae
1 changed files with 14 additions and 14 deletions
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28
res/res_pjsip.c
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@@ -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">