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-rw-r--r--configs/samples/sip.conf.sample32
1 files changed, 16 insertions, 16 deletions
diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample
index 75543036d..f8aa12f42 100644
--- a/configs/samples/sip.conf.sample
+++ b/configs/samples/sip.conf.sample
@@ -104,7 +104,7 @@
; ** Old configuration options **
; The "call-limit" configuation option is considered old is replaced
-; by new functionality. To enable callcounters, you use the new
+; by new functionality. To enable callcounters, you use the new
; "callcounter" setting (for extension states in queue and subscriptions)
; You are encouraged to use the dialplan groupcount functionality
; to enforce call limits instead of using this channel-specific method.
@@ -139,7 +139,7 @@ allowoverlap=no ; Disable overlap dialing support. (Default is y
; Set this to your host name or domain name
;domainsasrealm=no ; Use domains list as realms
; You can serve multiple Realms specifying several
- ; 'domain=...' directives (see below).
+ ; 'domain=...' directives (see below).
; In this case Realm will be based on request 'From'/'To' header
; and should match one of domain names.
; Otherwise default 'realm=...' will be used.
@@ -187,8 +187,8 @@ udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0
; makes the assumption that the endpoint supports all known SIP methods.
; If you know that your SIP endpoint does not provide support for a specific
; method, then you may provide a comma-separated list of methods that your
-; endpoint does not implement in the disallowed_methods option. Note that
-; if your endpoint is truthful with its Allow header, then there is no need
+; endpoint does not implement in the disallowed_methods option. Note that
+; if your endpoint is truthful with its Allow header, then there is no need
; to set this option. This option may be set in the general section or may
; be set per endpoint. If this option is set both in the general section and
; in a peer section, then the peer setting completely overrides the general
@@ -216,7 +216,7 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
; Remember that the IP address must match the common name (hostname) in the
; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
- ; For details how to construct a certificate for SIP see
+ ; For details how to construct a certificate for SIP see
; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
@@ -371,14 +371,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; but the remote party's domain will be anonymized. The way legacy
; behaves may violate RFC-3325, but it follows historic behavior.
; This option is set to 'legacy' by default
-;prematuremedia=no ; Some ISDN links send empty media frames before
- ; the call is in ringing or progress state. The SIP
+;prematuremedia=no ; Some ISDN links send empty media frames before
+ ; the call is in ringing or progress state. The SIP
; channel will then send 183 indicating early media
; which will be empty - thus users get no ring signal.
; Setting this to "yes" will stop any media before we have
; call progress (meaning the SIP channel will not send 183 Session
; Progress for early media). Default is "yes". Also make sure that
- ; the SIP peer is configured with progressinband=never.
+ ; the SIP peer is configured with progressinband=never.
;
; In order for "noanswer" applications to work, you need to run
; the progress() application in the priority before the app.
@@ -881,10 +881,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; externaddr = 12.34.56.78 ; use this address.
; externaddr = 12.34.56.78:9900 ; use this address and port.
; externaddr = mynat.my.org:12600 ; Public address of my nat box.
-; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
-; ; externtcpport will default to the externaddr or externhost port if either one is set.
+; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
+; ; externtcpport will default to the externaddr or externhost port if either one is set.
; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
-; ; externtlsport port will default to the RFC designated port of 5061.
+; ; externtlsport port will default to the RFC designated port of 5061.
;
; b. "externhost = hostname[:port]" is similar to "externaddr" except
; that the hostname is looked up every "externrefresh" seconds
@@ -1004,7 +1004,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; It only controls Asterisk generating reINVITEs for the specific
; purpose of setting up a direct media path. If a reINVITE is
; needed to switch a media stream to inactive (when placed on
- ; hold) or to T.38, it will still be done, regardless of this
+ ; hold) or to T.38, it will still be done, regardless of this
; setting. Note that direct T.38 is not supported.
;directmedia=nonat ; An additional option is to allow media path redirection
@@ -1031,11 +1031,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; callers INVITE. This will also fail if directmedia is enabled when
; the device is actually behind NAT.
-;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
+;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
; (There is no default setting, this is just an example)
; Use this if some of your phones are on IP addresses that
- ; can not reach each other directly. This way you can force
+ ; can not reach each other directly. This way you can force
; RTP to always flow through asterisk in such cases.
;directmediaacl=acl_example ; Use named ACLs defined in acl.conf
@@ -1220,10 +1220,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; matches rather than the IP address. Since it is also a peer, a friend entity can
; be called as long as its IP is known to Asterisk. In the case of host=dynamic,
; this means it is necessary for the entity to register before Asterisk can call it.
-;
+;
; Use remotesecret for outbound authentication, and secret for authenticating
; inbound requests. For historical reasons, if no remotesecret is supplied for an
-; outbound registration or call, the secret will be used.
+; outbound registration or call, the secret will be used.
;
; For device names, we recommend using only a-z, numerics (0-9) and underscore
;