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authorRichard Mudgett <rmudgett@digium.com>2016-07-15 16:16:18 -0500
committerRichard Mudgett <rmudgett@digium.com>2016-07-21 23:30:57 -0500
commit33716106e00da12d24999610aea60a76cc5bbdb5 (patch)
treeb1c326a54a97da5d7392f518b2c5839b08c93e10 /res/res_pjsip.c
parent80a989910077e0df020d754252c32cfbb8c5578f (diff)
res_pjsip: Whitespace and comment cleanup.
Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38
Diffstat (limited to 'res/res_pjsip.c')
-rw-r--r--res/res_pjsip.c51
1 files changed, 25 insertions, 26 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 60b8252ad..3870e9f8d 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -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
@@ -811,7 +810,7 @@
<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>
@@ -1452,9 +1451,9 @@
<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>
@@ -1463,15 +1462,15 @@
</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
@@ -1485,17 +1484,17 @@
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>
@@ -2060,7 +2059,7 @@
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>
@@ -2096,7 +2095,7 @@
<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>