From 0f9b144c1aaf7a68050051dcb069923bc422f4df Mon Sep 17 00:00:00 2001 From: Richard Mudgett Date: Wed, 31 Aug 2016 15:56:41 -0500 Subject: Sample configs: Eliminate false multiline comment block starts. Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6 --- configs/samples/alsa.conf.sample | 4 ++-- configs/samples/ccss.conf.sample | 16 ++++++------- configs/samples/chan_dahdi.conf.sample | 4 ++-- configs/samples/console.conf.sample | 4 ++-- configs/samples/mgcp.conf.sample | 6 ++--- configs/samples/minivm.conf.sample | 14 +++++------ configs/samples/misdn.conf.sample | 4 ++-- configs/samples/oss.conf.sample | 4 ++-- configs/samples/queues.conf.sample | 4 ++-- configs/samples/res_snmp.conf.sample | 2 +- configs/samples/sip.conf.sample | 44 +++++++++++++++++----------------- configs/samples/skinny.conf.sample | 20 ++++++++-------- configs/samples/unistim.conf.sample | 4 ++-- configs/samples/vpb.conf.sample | 2 +- 14 files changed, 66 insertions(+), 66 deletions(-) (limited to 'configs') 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= ; 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= ; 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 -- cgit v1.2.3