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-rw-r--r--configs/sip.conf.sample6
1 files changed, 3 insertions, 3 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 7cf083761..1b16543ef 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -59,7 +59,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;tos=184 ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpiry=3600 ; Max length of incoming registration we allow
-;defaultexpiry=120 ; Default length of incoming/outoing registration
+;defaultexpiry=120 ; Default length of incoming/outgoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
@@ -338,7 +338,7 @@ 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
@@ -428,7 +428,7 @@ 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
;[cisco1]
;type=friend