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-rw-r--r--configs/sip.conf.sample12
1 files changed, 6 insertions, 6 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index b04c60e77..f931780b9 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -60,7 +60,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
; and subscriptions (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
-;defaultexpiry=120 ; Default length of incoming/outoing registration
+;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
; Defaults to 100 ms
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
@@ -294,7 +294,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; For non-realtime peers, when their registration expires, the
; information will _not_ be removed from memory or the Asterisk database
; if you attempt to place a call to the peer, the existing information
- ; will be used in spiteof it having expired
+ ; will be used in spite of it having expired
;
; For realtime peers, when the peer is retrieved from realtime storage,
; the registration information will be used regardless of whether
@@ -353,11 +353,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
- ; big jumps in/broken timestamps, usualy sent from exotic devices
+ ; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
- ; channel. Two implementations are currenlty available - "fixed"
+ ; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
@@ -465,7 +465,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
; For local phones, type=friend works most of the time
;
-; If you have one-way audio, you propably have NAT problems.
+; If you have one-way audio, you probably have NAT problems.
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
@@ -566,7 +566,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
;callgroup=1,3-4 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
-;defaultip=192.168.0.60 ; IP address to use if peer has not registred
+;defaultip=192.168.0.60 ; IP address to use if peer has not registered
;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
;permit=192.168.0.60/255.255.255.0