Sample configs: Eliminate false multiline comment block starts.

Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6

14 files changed, 66 insertions, 66 deletions

diff --git a/configs/samples/alsa.conf.sample b/configs/samples/alsa.conf.sample
index ced5b4485..23aac4e10 100644
--- a/configs/samples/alsa.conf.sample
+++ b/configs/samples/alsa.conf.sample
@@ -46,7 +46,7 @@ extension=s
 ; 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 @@ extension=s
                               ; 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 21b0b0668..7b3fe7d23 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 6dd365f47..d0ccd5db5 100644
--- a/configs/samples/chan_dahdi.conf.sample
+++ b/configs/samples/chan_dahdi.conf.sample
@@ -1220,7 +1220,7 @@ pickupgroup=1
 ;
 ;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 @@ pickupgroup=1
                               ; 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 606254eee..aad306ed5 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 7c725bc3d..f4bc0dbf2 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 @@ matchdigittimeout = 5000  ; defaults 3000 = 3s
                               ; 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 55a39c869..2df3449d1 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 @@ silencethreshold=128
 ; 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 @@ emaildateformat=%A, %B %d, %Y at %r
 ; 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 @@ emaildateformat=%A, %B %d, %Y at %r
 ;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 @@ central=America/Chicago|'vm-received' Q 'digits/at' IMp
 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@digium.com
@@ -187,7 +187,7 @@ dateformat=%A, %B %d, %Y at %r
 ;subject = Dear old chap, you've got an electronic communique
 ;charset=ascii
 
-;----------------------- Mailbox accounts --------------------------
+; ---------------------- Mailbox accounts --------------------------
 ;Template for mailbox definition - all options
 ;
 ;	[username@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 ac54dbc5a..ca27c03bd 100644
--- a/configs/samples/misdn.conf.sample
+++ b/configs/samples/misdn.conf.sample
@@ -109,7 +109,7 @@ crypt_prefix=**
 ;
 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 @@ crypt_keys=test,muh
                               ; 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 c3781a2a2..ee169209f 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 85cf9e40f..8a9c88402 100644
--- a/configs/samples/queues.conf.sample
+++ b/configs/samples/queues.conf.sample
@@ -129,7 +129,7 @@ monitor-type = MixMonitor
 ;
 ;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 @@ monitor-type = MixMonitor
 ;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 a6e40c8e2..7f3734910 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 27012614e..c5ffdcccd 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                            ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; 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@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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;
 ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 				; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 
 ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;						; 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 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;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 be88dc230..2bf06fbc8 100644
--- a/configs/samples/skinny.conf.sample
+++ b/configs/samples/skinny.conf.sample
@@ -54,7 +54,7 @@ keepalive=120
 ;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 @@ keepalive=120
                              ; 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 @@ keepalive=120
 ; 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 @@ keepalive=120
 
 
 [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 @@ keepalive=120
 ; 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 c33426b0c..a09642796 100644
--- a/configs/samples/unistim.conf.sample
+++ b/configs/samples/unistim.conf.sample
@@ -17,7 +17,7 @@ port=5000                    ; UDP port
 ;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 @@ port=5000                    ; UDP port
                               ; 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 fecb3ec59..bdc89dff5 100644
--- a/configs/samples/vpb.conf.sample
+++ b/configs/samples/vpb.conf.sample
@@ -199,7 +199,7 @@ grunttimeout=3600
 ;
 mode=immediate
 
-;-------------------------------------------------------------------------
+; ------------------------------------------------------------------------
 ; Channel definitions
 ;
 ; Each channel inherits the settings specified above, unless the are