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Sample configs: Eliminate false multiline comment block starts.

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Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6
This commit is contained in:
Richard Mudgett
2016-08-31 15:56:41 -05:00
parent d3c4b901d4
commit 4aaa27e532
14 changed files with 66 additions and 66 deletions
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44
configs/samples/sip.conf.sample
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@@ -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
@@ -87,7 +87,7 @@
; 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.
@@ -111,7 +111,7 @@
; 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
@@ -573,7 +573,7 @@ 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. If the
@@ -622,7 +622,7 @@ 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.
@@ -636,7 +636,7 @@ 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
@@ -652,7 +652,7 @@ 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.
@@ -681,7 +681,7 @@ 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
@@ -692,7 +692,7 @@ 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
@@ -741,7 +741,7 @@ 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.
@@ -774,7 +774,7 @@ 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]
@@ -851,7 +851,7 @@ 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:
@@ -866,7 +866,7 @@ 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.
@@ -1008,7 +1008,7 @@ 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
@@ -1090,7 +1090,7 @@ 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
;
@@ -1128,7 +1128,7 @@ 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
@@ -1167,13 +1167,13 @@ 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
@@ -1205,7 +1205,7 @@ 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
@@ -1224,7 +1224,7 @@ 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.
@@ -1351,7 +1351,7 @@ 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.
@@ -1409,7 +1409,7 @@ 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