[Asterisk-code-review] Sample configs: Eliminate false multiline comment block starts. (asterisk[13])
Anonymous Coward
asteriskteam at digium.com
Tue Sep 6 12:24:17 CDT 2016
Anonymous Coward #1000019 has submitted this change and it was merged.
Change subject: Sample configs: Eliminate false multiline comment block starts.
......................................................................
Sample configs: Eliminate false multiline comment block starts.
Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6
---
M configs/samples/alsa.conf.sample
M configs/samples/ccss.conf.sample
M configs/samples/chan_dahdi.conf.sample
M configs/samples/console.conf.sample
M configs/samples/mgcp.conf.sample
M configs/samples/minivm.conf.sample
M configs/samples/misdn.conf.sample
M configs/samples/oss.conf.sample
M configs/samples/queues.conf.sample
M configs/samples/res_snmp.conf.sample
M configs/samples/sip.conf.sample
M configs/samples/skinny.conf.sample
M configs/samples/unistim.conf.sample
M configs/samples/vpb.conf.sample
14 files changed, 66 insertions(+), 66 deletions(-)
Approvals:
George Joseph: Looks good to me, approved
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/configs/samples/alsa.conf.sample b/configs/samples/alsa.conf.sample
index ced5b44..23aac4e 100644
--- a/configs/samples/alsa.conf.sample
+++ b/configs/samples/alsa.conf.sample
@@ -46,7 +46,7 @@
; systems where there will be no return audio path, such as overhead pagers.
;noaudiocapture=true
-;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; ALSA channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
@@ -74,5 +74,5 @@
; network normally has low jitter, but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
-;-----------------------------------------------------------------------------------
+; ----------------------------------------------------------------------------------
diff --git a/configs/samples/ccss.conf.sample b/configs/samples/ccss.conf.sample
index 21b0b06..7b3fe7d 100644
--- a/configs/samples/ccss.conf.sample
+++ b/configs/samples/ccss.conf.sample
@@ -64,9 +64,9 @@
; PLEASE READ THIS!!!
;===========================================
;
-;---------------------------------------------------------------------
+; --------------------------------------------------------------------
; Timers
-;---------------------------------------------------------------------
+; --------------------------------------------------------------------
;There are three configurable timers for all types of CC: the
;cc_offer_timer, the ccbs_available_timer, and the ccnr_available_timer.
;In addition, when using a generic agent, there is a fourth timer,
@@ -98,9 +98,9 @@
; only affects operation when using a generic agent.
;
;cc_recall_timer = 20
-;---------------------------------------------------------------------
+; --------------------------------------------------------------------
; Policies
-;---------------------------------------------------------------------
+; --------------------------------------------------------------------
; Policy settings tell Asterisk how to behave and what sort of
; resources to allocate in order to facilitate CC. There are two
; settings to control the actions Asterisk will take.
@@ -153,9 +153,9 @@
;cc_monitor_policy=never
;
;
-;---------------------------------------------------------------------
+; --------------------------------------------------------------------
; Limits
-;---------------------------------------------------------------------
+; --------------------------------------------------------------------
;
; The use of CC requires Asterisk to potentially use more memory than
; some administrators would like. As such, it is a good idea to limit
@@ -175,9 +175,9 @@
;
;cc_max_monitors = 5
;
-;---------------------------------------------------------------------
+; --------------------------------------------------------------------
; Other
-;---------------------------------------------------------------------
+; --------------------------------------------------------------------
;
; When using a generic CC agent, the caller who requested CC will be
; called back when a called party becomes available. When the caller
diff --git a/configs/samples/chan_dahdi.conf.sample b/configs/samples/chan_dahdi.conf.sample
index 6dd365f..d0ccd5d 100644
--- a/configs/samples/chan_dahdi.conf.sample
+++ b/configs/samples/chan_dahdi.conf.sample
@@ -1220,7 +1220,7 @@
;
;jitterbuffers=4
;
-;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
@@ -1248,7 +1248,7 @@
; network normally has low jitter, but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
-;-----------------------------------------------------------------------------------
+; ----------------------------------------------------------------------------------
;
; You can define your own custom ring cadences here. You can define up to 8
; pairs. If the silence is negative, it indicates where the caller ID spill is
diff --git a/configs/samples/console.conf.sample b/configs/samples/console.conf.sample
index 606254e..aad306e 100644
--- a/configs/samples/console.conf.sample
+++ b/configs/samples/console.conf.sample
@@ -44,7 +44,7 @@
;
;mohinterpret=default
-;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; Console channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
@@ -72,7 +72,7 @@
; network normally has low jitter, but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
-;-----------------------------------------------------------------------------------
+; ----------------------------------------------------------------------------------
;
diff --git a/configs/samples/mgcp.conf.sample b/configs/samples/mgcp.conf.sample
index 7c725bc..f4bc0db 100644
--- a/configs/samples/mgcp.conf.sample
+++ b/configs/samples/mgcp.conf.sample
@@ -11,12 +11,12 @@
;cos=3 ; Sets 802.1p priority for signaling packets.
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
-;---------------------- DIGIT TIMEOUTS ----------------------------
+; --------------------- DIGIT TIMEOUTS ----------------------------
firstdigittimeout = 30000 ; default 16000 = 16s
gendigittimeout = 10000 ; default 8000 = 8s
matchdigittimeout = 5000 ; defaults 3000 = 3s
-;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; MGCP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
@@ -48,7 +48,7 @@
; network normally has low jitter, but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
-;-----------------------------------------------------------------------------------
+; ----------------------------------------------------------------------------------
;[dlinkgw]
;host = 192.168.0.64
diff --git a/configs/samples/minivm.conf.sample b/configs/samples/minivm.conf.sample
index 55a39c8..2df3449 100644
--- a/configs/samples/minivm.conf.sample
+++ b/configs/samples/minivm.conf.sample
@@ -12,7 +12,7 @@
; this configuration file or realtime. The idea is to build voicemail as building blocks so that
; a complete and adaptive voicemail system can be built in the dialplan
;
-;------------------------------ Variables to use in subject, from and message body ------------------
+; ----------------------------- Variables to use in subject, from and message body ------------------
; Change the from, body and/or subject, variables:
; MVM_NAME, MVM_DUR, MVM_MSGNUM, VM_MAILBOX, MVM_CALLERID, MVM_CIDNUM,
; MVM_CIDNAME, MVM_DATE
@@ -24,7 +24,7 @@
; Note: The emailbody config row can only be up to 512 characters due to a
; limitation in the Asterisk configuration subsystem.
; To create longer mails, use the templatefile option when creating the template
-;----------------------------------------------------------------------------------------------------
+; ---------------------------------------------------------------------------------------------------
[general]
; Default format for storing and sending voicemail
@@ -64,7 +64,7 @@
; This is used both for e-mail and pager messages
;mailcmd=/usr/sbin/sendmail -t
;
-;--------------Default e-mail message template (used if no templates are used) ------
+; -------------Default e-mail message template (used if no templates are used) ------
;fromstring=The Asterisk PBX
;
@@ -82,7 +82,7 @@
; 24h date format
;emaildateformat=%A, %d %B %Y at %H:%M:%S
;
-;--------------Default pager message template (used if no templates are used) ------
+; -------------Default pager message template (used if no templates are used) ------
; You can also change the Pager From: string, the pager body and/or subject.
; The above defined variables also can be used here
;pagerfromstring=The Asterisk PBX
@@ -90,7 +90,7 @@
;pagerbody=New ${MVM_DUR} long msg in box ${MVM_MAILBOX}\nfrom ${MVM_CALLERID}, on ${MVM_DATE}
;
;
-;--------------Timezone definitions (used in voicemail accounts) -------------------
+; -------------Timezone definitions (used in voicemail accounts) -------------------
;
; Users may be located in different timezones, or may have different
; message announcements for their introductory message when they enter
@@ -133,7 +133,7 @@
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
-;----------------------- Message body templates---------------------
+; ---------------------- Message body templates---------------------
; [template-name] ; "template-" is a verbatim marker
; fromaddress = Your Friendly Asterisk Server
; fromemail = asteriskvm at digium.com
@@ -187,7 +187,7 @@
;subject = Dear old chap, you've got an electronic communique
;charset=ascii
-;----------------------- Mailbox accounts --------------------------
+; ---------------------- Mailbox accounts --------------------------
;Template for mailbox definition - all options
;
; [username at domain] ; Has to be unique within domain (MWM_USERNAME, MWM_DOMAIN)
diff --git a/configs/samples/misdn.conf.sample b/configs/samples/misdn.conf.sample
index ac54dbc..ca27c03 100644
--- a/configs/samples/misdn.conf.sample
+++ b/configs/samples/misdn.conf.sample
@@ -109,7 +109,7 @@
;
crypt_keys=test,muh
-;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
@@ -140,7 +140,7 @@
; network normally has low jitter, but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
-;-----------------------------------------------------------------------------------
+; ----------------------------------------------------------------------------------
; users sections:
;
diff --git a/configs/samples/oss.conf.sample b/configs/samples/oss.conf.sample
index c3781a2..ee16920 100644
--- a/configs/samples/oss.conf.sample
+++ b/configs/samples/oss.conf.sample
@@ -46,7 +46,7 @@
; queuesize = 10 ; frames in device driver
; frags = 8 ; argument to SETFRAGMENT
- ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+ ; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; OSS channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
@@ -74,7 +74,7 @@
; network normally has low jitter, but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
- ;-----------------------------------------------------------------------------------
+ ; ----------------------------------------------------------------------------------
; below is an entry for a second console channel
; [card1]
diff --git a/configs/samples/queues.conf.sample b/configs/samples/queues.conf.sample
index 85cf9e4..8a9c884 100644
--- a/configs/samples/queues.conf.sample
+++ b/configs/samples/queues.conf.sample
@@ -129,7 +129,7 @@
;
;penaltymemberslimit = 5
;
-;----------------------QUEUE TIMING OPTIONS------------------------------------
+; ---------------------QUEUE TIMING OPTIONS------------------------------------
; A Queue has two different "timeout" values associated with it. One is the
; timeout parameter configured in queues.conf. This timeout specifies the
; amount of time to try ringing a member's phone before considering the
@@ -181,7 +181,7 @@
;retry = 5
;timeoutpriority = app|conf
;
-;-----------------------END QUEUE TIMING OPTIONS---------------------------------
+; ----------------------END QUEUE TIMING OPTIONS---------------------------------
; Weight of queue - when compared to other queues, higher weights get
; first shot at available channels when the same channel is included in
; more than one queue.
diff --git a/configs/samples/res_snmp.conf.sample b/configs/samples/res_snmp.conf.sample
index a6e40c8..7f37349 100644
--- a/configs/samples/res_snmp.conf.sample
+++ b/configs/samples/res_snmp.conf.sample
@@ -1,6 +1,6 @@
;
; Configuration file for res_snmp
-;---------------------------------
+; --------------------------------
;
; Res_snmp can run as a subagent or standalone SNMP agent. The standalone snmp
; agent is based on net-snmp and will read a configuration file called
diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample
index 2701261..c5ffdcc 100644
--- a/configs/samples/sip.conf.sample
+++ b/configs/samples/sip.conf.sample
@@ -15,7 +15,7 @@
; - context - Which set of services you offer various users
;
; SIP dial strings
-;-----------------------------------------------------------
+; ----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
; SIP/devicename
@@ -76,7 +76,7 @@
; sip reload Reload configuration file
; sip show settings Show the current channel configuration
;
-;------- Naming devices ------------------------------------------------------
+; ------ Naming devices ------------------------------------------------------
;
; When naming devices, make sure you understand how Asterisk matches calls
; that come in.
@@ -100,7 +100,7 @@
; not needed at all. Check below. In later releases, it's renamed
; to "defaultuser" which is a better name, since it is used in
; combination with the "defaultip" setting.
-;-----------------------------------------------------------------------------
+; ----------------------------------------------------------------------------
; ** Old configuration options **
; The "call-limit" configuation option is considered old is replaced
@@ -559,7 +559,7 @@
; are not purged during SIP reloads.
;
-;------------------------ TLS settings ------------------------------------------------------------
+; ----------------------- TLS settings ------------------------------------------------------------
;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
; The certificates must be sorted starting with the subject's certificate
; and followed by intermediate CA certificates if applicable.
@@ -603,7 +603,7 @@
; Your distribution might have changed that list
; further.
;
-;--------------------------- SIP timers ----------------------------------------------------
+; -------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
@@ -617,7 +617,7 @@
; in this amount of time, the call will autocongest
; Defaults to 64*timert1
-;--------------------------- RTP timers ----------------------------------------------------
+; -------------------------- RTP timers ----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
@@ -633,7 +633,7 @@
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
; (default is off - zero)
-;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
+; -------------------------- SIP Session-Timers (RFC 4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
; This mechanism can detect and reclaim SIP channels that do not terminate through normal
; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
@@ -662,7 +662,7 @@
;session-minse=90
;session-refresher=uac
;
-;--------------------------- SIP DEBUGGING ---------------------------------------------------
+; -------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration.
; NOTE: You cannot use the CLI to turn it off. You'll
@@ -673,7 +673,7 @@
; SIP history is output to the DEBUG logging channel
-;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
+; -------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
; You can subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE
@@ -718,7 +718,7 @@
;callcounter = yes ; Enable call counters on devices. This can be set per
; device too.
-;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
+; ---------------------------------------- T.38 FAX SUPPORT ----------------------------------
;
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
@@ -751,7 +751,7 @@
; faxdetect = cng ; Enables only CNG detection
; faxdetect = t38 ; Enables only T.38 detection
;
-;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
+; ---------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
@@ -828,7 +828,7 @@
; 401 responses and continue retrying according to normal
; retry rules.
-;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
+; ---------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
; by other phones. At this time, you can only subscribe using UDP as the transport.
; Format for the mwi register statement is:
@@ -843,7 +843,7 @@
; MWI received will be stored in the 1234 mailbox of the SIP_Remote context.
; It can be used by other phones by following the below:
; mailbox=1234 at SIP_Remote
-;----------------------------------------- NAT SUPPORT ------------------------
+; ---------------------------------------- NAT SUPPORT ------------------------
;
; WARNING: SIP operation behind a NAT is tricky and you really need
; to read and understand well the following section.
@@ -981,7 +981,7 @@
;
; icesupport = yes
-;----------------------------------- MEDIA HANDLING --------------------------------
+; ---------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
; no reason for Asterisk to stay in the media path, the media will be redirected.
; This does not really work well in the case where Asterisk is outside and the
@@ -1063,7 +1063,7 @@
; option may be specified at the global or peer scope.
;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
; media streams when appropriate, even if a DTLS stream is present.
-;----------------------------------------- REALTIME SUPPORT ------------------------
+; ---------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
;
@@ -1101,7 +1101,7 @@
; is still in memory (due to caching or other reasons), the
; information will not be removed from realtime storage
-;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
+; ---------------------------------------- SIP DOMAIN SUPPORT ------------------------
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
; domains, each of which can direct the call to a specific context if desired.
; By default, all domains are accepted and sent to the default context or the
@@ -1140,13 +1140,13 @@
; destinations which do not have a prior
; account relationship with your server.
-;------------------------------ Advice of Charge CONFIGURATION --------------------------
+; ----------------------------- Advice of Charge CONFIGURATION --------------------------
; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
; AOC-E to snom endpoints. This option can be used both in the
; peer and global scope. The default for this option is off.
-;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
@@ -1178,7 +1178,7 @@
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
-;-----------------------------------------------------------------------------------
+; ----------------------------------------------------------------------------------
[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
@@ -1197,7 +1197,7 @@
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm
-;------------------------------------------------------------------------------
+; -----------------------------------------------------------------------------
; DEVICE CONFIGURATION
;
; SIP entities have a 'type' which determines their roles within Asterisk.
@@ -1324,7 +1324,7 @@
; ; from the peer's configuration.
;
-;------------------------------------------------------------------------------
+; -----------------------------------------------------------------------------
; DTLS-SRTP CONFIGURATION
;
; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
@@ -1379,7 +1379,7 @@
;port=80 ; The port number we want to connect to on the remote side
; Also used as "defaultport" in combination with "defaultip" settings
-;--- sample definition for a provider
+; -- sample definition for a provider
;[provider1]
;type=peer
;host=sip.provider1.com
diff --git a/configs/samples/skinny.conf.sample b/configs/samples/skinny.conf.sample
index be88dc2..2bf06fb 100644
--- a/configs/samples/skinny.conf.sample
+++ b/configs/samples/skinny.conf.sample
@@ -54,7 +54,7 @@
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
;cos_video=4 ; Sets 802.1p priority for RTP video packets.
-;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; skinny channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
@@ -79,10 +79,10 @@
; Defaults to fixed.
;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
-;-----------------------------------------------------------------------------------
+; ----------------------------------------------------------------------------------
[lines]
-;----------------------------------- LINES SECTION --------------------------------
+; ---------------------------------- LINES SECTION --------------------------------
; Options set under [lines] apply to all lines unless explicitly set for a particular
; device. The options that can be set under lines are specified in GENERAL LINE OPTIONS.
; These options can also be set for each individual device as well as those under SPECIFIC
@@ -95,15 +95,15 @@
; Where options are common to both lines and devices, the results typically take that of
; the least permission. ie if a no is set for either line or device, the call will not be
; able to use that permission
-;-------------------------------- GENERAL LINE OPTIONS -----------------------------
+; ------------------------------- GENERAL LINE OPTIONS -----------------------------
;earlyrtp=1 ; whether audio signalling should be provided by asterisk
; ; (earlyrtp=1) or device generated (earlyrtp=0). default=yes
;transfer=1 ; whether the device is allowed to transfer. default=yes
;context=default ; context to use for this line.
;callfwdtimeout=20000 ; ms before cfwd_noans occurs (default 20 secs)
-;------------------------------- SPECIFIC LINE OPTIONS -----------------------------
+; ------------------------------ SPECIFIC LINE OPTIONS -----------------------------
;setvar= ; allows for the setting of chanvars.
-;-----------------------------------------------------------------------------------
+; ----------------------------------------------------------------------------------
;[100]
;nat=yes
@@ -149,7 +149,7 @@
[devices]
-;---------------------------------- DEVICES SECTION -------------------------------
+; --------------------------------- DEVICES SECTION -------------------------------
; Options set under [devices] apply to all devices unless explicitly set for a particular
; device. The options that can be set under devices are specified in GENERAL DEVICE OPTIONS.
; These options can also be set for each individual device as well as those under SPECIFIC
@@ -162,16 +162,16 @@
; Where options are common to both lines and devices, the results typically take that of
; the least permission. ie if a no is set for either line or device, the call will not be
; able to use that permission
-;------------------------------- GENERAL DEVICE OPTIONS ----------------------------
+; ------------------------------ GENERAL DEVICE OPTIONS ----------------------------
;earlyrtp=1 ; whether audio signalling should be provided by asterisk
; ; (earlyrtp=1) or device generated (earlyrtp=0). default=yes
;transfer=1 ; whether the device is allowed to transfer. default=yes
-;------------------------------ SPECIFIC DEVICE OPTIONS ----------------------------
+; ----------------------------- SPECIFIC DEVICE OPTIONS ----------------------------
;device="SEPxxxxxxxxxxxx ; id of the device. Must be set.
;version=P002G204 ; firmware version to be loaded. If this version is different
; ; to the one on the device, the device will try to load this
; ; version from the tftp server. Set to device firmware version.
-;-----------------------------------------------------------------------------------
+; ----------------------------------------------------------------------------------
; Typical config for 12SP+
;[florian]
diff --git a/configs/samples/unistim.conf.sample b/configs/samples/unistim.conf.sample
index c33426b..a096427 100644
--- a/configs/samples/unistim.conf.sample
+++ b/configs/samples/unistim.conf.sample
@@ -17,7 +17,7 @@
;autoprovisioning=no ; Allow undeclared phones to register an extension. See README for important
; informations. no (default), yes, tn.
;mohsuggest=default
-;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
@@ -41,7 +41,7 @@
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
-;-----------------------------------------------------------------------------------
+; ----------------------------------------------------------------------------------
;[black] ; name of the device
diff --git a/configs/samples/vpb.conf.sample b/configs/samples/vpb.conf.sample
index fecb3ec..bdc89df 100644
--- a/configs/samples/vpb.conf.sample
+++ b/configs/samples/vpb.conf.sample
@@ -199,7 +199,7 @@
;
mode=immediate
-;-------------------------------------------------------------------------
+; ------------------------------------------------------------------------
; Channel definitions
;
; Each channel inherits the settings specified above, unless the are
--
To view, visit https://gerrit.asterisk.org/3789
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
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