[asterisk-commits] res pjsip: Whitespace and comment cleanup. (asterisk[14])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jul 22 07:45:21 CDT 2016
Anonymous Coward #1000019 has submitted this change and it was merged.
Change subject: res_pjsip: Whitespace and comment cleanup.
......................................................................
res_pjsip: Whitespace and comment cleanup.
Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38
---
M configs/samples/pjsip.conf.sample
M include/asterisk/res_pjsip.h
M res/res_pjsip.c
3 files changed, 36 insertions(+), 37 deletions(-)
Approvals:
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, approved
Corey Farrell: Looks good to me, but someone else must approve
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 9069a66..5fd7941 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -672,7 +672,7 @@
; usage of media encryption for this endpoint (default:
; "no")
;media_encryption_optimistic=no ; Use encryption if possible but don't fail the call
- ; if not possible.
+ ; if not possible.
;g726_non_standard=no ; When set to "yes" and an endpoint negotiates g.726
; audio then g.726 for AAL2 packing order is used contrary
; to what is recommended in RFC3551. Note, 'g726aal2' also
@@ -748,7 +748,7 @@
;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80
; byte tags (default: "no")
;set_var= ; Variable set on a channel involving the endpoint. For multiple
- ; channel variables specify multiple 'set_var'(s)
+ ; channel variables specify multiple 'set_var'(s)
;rtp_keepalive= ; Interval, in seconds, between comfort noise RTP packets if
; RTP is not flowing. This setting is useful for ensuring that
; holes in NATs and firewalls are kept open throughout a call.
@@ -790,7 +790,7 @@
; (default: "")
;ca_list_path= ; Path to directory containing certificates to read TLS ONLY.
; PJProject version 2.4 or higher is required for this option to
- ; be used.
+ ; be used.
; (default: "")
;cert_file= ; Certificate file for endpoint TLS ONLY
; Will read .crt or .pem file but only uses cert,
@@ -882,8 +882,8 @@
;disable_tcp_switch=yes ; Disable automatic switching from UDP to TCP transports
; if outgoing request is too large.
; See RFC 3261 section 18.1.1.
- ; Disabling this option has been known to cause interoperability
- ; issues, so disable at your own risk.
+ ; Disabling this option has been known to cause interoperability
+ ; issues, so disable at your own risk.
; (default: "yes")
;type= ; Must be of type system (default: "")
@@ -913,10 +913,10 @@
;contact_expiration_check_interval=30
; The interval (in seconds) to check for expired contacts.
;disable_multi_domain=no
- ; Disable Multi Domain support.
- ; If disabled it can improve realtime performace by reducing
- ; number of database requsts
- ; (default: "no")
+ ; Disable Multi Domain support.
+ ; If disabled it can improve realtime performace by reducing
+ ; number of database requsts
+ ; (default: "no")
;endpoint_identifier_order=ip,username,anonymous
; The order by which endpoint identifiers are given priority.
; Currently, "ip", "username", "auth_username" and "anonymous" are valid
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 6f59283..594259b 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -749,9 +749,9 @@
unsigned int usereqphone;
/*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */
unsigned int moh_passthrough;
- /* Access control list */
+ /*! Access control list */
struct ast_acl_list *acl;
- /* Restrict what IPs are allowed in the Contact header (for registration) */
+ /*! Restrict what IPs are allowed in the Contact header (for registration) */
struct ast_acl_list *contact_acl;
};
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 882f578..09581bc 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
@@ -806,7 +805,7 @@
<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>
@@ -1447,9 +1446,9 @@
<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>
@@ -1458,15 +1457,15 @@
</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
@@ -1480,17 +1479,17 @@
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>
@@ -2058,7 +2057,7 @@
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>
@@ -2094,7 +2093,7 @@
<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>
--
To view, visit https://gerrit.asterisk.org/3302
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Gerrit-MessageType: merged
Gerrit-Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Corey Farrell <git at cfware.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
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