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res_pjsip: Whitespace and comment cleanup.

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Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38
This commit is contained in:
Richard Mudgett
2016-07-15 16:16:18 -05:00
parent 9473818659
commit 4286a369a1
3 changed files with 36 additions and 37 deletions
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51
res/res_pjsip.c
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@@ -217,10 +217,9 @@
<enum name="info">
<para>DTMF is sent as SIP INFO packets.</para>
</enum>
<enum name="auto">
<para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
</enum>
<enum name="auto">
<para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
</enum>
</enumlist>
</description>
</configOption>
@@ -510,15 +509,15 @@
<configOption name="g726_non_standard" default="no">
<synopsis>Force g.726 to use AAL2 packing order when negotiating g.726 audio</synopsis>
<description><para>
When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
packing order instead of what is recommended by RFC3551. Since this essentially
replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
specified in the endpoint's allowed codec list.
When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2
packing order instead of what is recommended by RFC3551. Since this essentially
replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be
specified in the endpoint's allowed codec list.
</para></description>
</configOption>
<configOption name="inband_progress" default="no">
<synopsis>Determines whether chan_pjsip will indicate ringing using inband
progress.</synopsis>
progress.</synopsis>
<description><para>
If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
when told to indicate ringing and will immediately start sending ringing
@@ -814,7 +813,7 @@
<configOption name="set_var">
<synopsis>Variable set on a channel involving the endpoint.</synopsis>
<description><para>
When a new channel is created using the endpoint set the specified
When a new channel is created using the endpoint set the specified
variable(s) on that channel. For multiple channel variables specify
multiple 'set_var'(s).
</para></description>
@@ -1455,9 +1454,9 @@
<synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis>
</configOption>
<configOption name="regcontext" default="">
<synopsis>When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given
peer who registers or unregisters with us.</synopsis>
</configOption>
<synopsis>When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given
peer who registers or unregisters with us.</synopsis>
</configOption>
<configOption name="default_outbound_endpoint" default="default_outbound_endpoint">
<synopsis>Endpoint to use when sending an outbound request to a URI without a specified endpoint.</synopsis>
</configOption>
@@ -1466,15 +1465,15 @@
</configOption>
<configOption name="debug" default="no">
<synopsis>Enable/Disable SIP debug logging. Valid options include yes|no or
a host address</synopsis>
a host address</synopsis>
</configOption>
<configOption name="endpoint_identifier_order" default="ip,username,anonymous">
<synopsis>The order by which endpoint identifiers are processed and checked.
Identifier names are usually derived from and can be found in the endpoint
identifier module itself (res_pjsip_endpoint_identifier_*).
You can use the CLI command "pjsip show identifiers" to see the
identifiers currently available.</synopsis>
<description>
Identifier names are usually derived from and can be found in the endpoint
identifier module itself (res_pjsip_endpoint_identifier_*).
You can use the CLI command "pjsip show identifiers" to see the
identifiers currently available.</synopsis>
<description>
<note><para>
One of the identifiers is "auth_username" which matches on the username in
an Authentication header. This method has some security considerations because an
@@ -1488,17 +1487,17 @@
how many unmatched requests are received from a single ip address before a security
event is generated using the unidentified_request parameters.
</para></note>
</description>
</description>
</configOption>
<configOption name="default_from_user" default="asterisk">
<synopsis>When Asterisk generates an outgoing SIP request, the From header username will be
set to this value if there is no better option (such as CallerID) to be
used.</synopsis>
set to this value if there is no better option (such as CallerID) to be
used.</synopsis>
</configOption>
<configOption name="default_realm" default="asterisk">
<synopsis>When Asterisk generates an challenge, the digest will be
set to this value if there is no better option (such as auth/realm) to be
used.</synopsis>
set to this value if there is no better option (such as auth/realm) to be
used.</synopsis>
</configOption>
</configObject>
</configFile>
@@ -2066,7 +2065,7 @@
Provides a listing of all endpoints. For each endpoint an <literal>EndpointList</literal> event
is raised that contains relevant attributes and status information. Once all
endpoints have been listed an <literal>EndpointListComplete</literal> event is issued.
</para>
</para>
</description>
<responses>
<list-elements>
@@ -2102,7 +2101,7 @@
<literal>IdentifyDetail</literal>. Some events may be listed multiple times if multiple objects are
associated (for instance AoRs). Once all detail events have been raised a final
<literal>EndpointDetailComplete</literal> event is issued.
</para>
</para>
</description>
<responses>
<list-elements>