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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2006-01-24 18:15:20 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2006-01-24 18:15:20 +0000
commit295fbee42e2b6507348cce689e0eeb6509f15aa5 (patch)
tree7bc47931ae4a5f8a8764a0950515364295ae0420 /configs
parent353cec56ea1dea09a99bb960ebdd1f96a209190f (diff)
Make it clear that caller ID in sip.conf is used only on incoming calls (inspired by bug #6183)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8554 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rw-r--r--configs/sip.conf.sample1
1 files changed, 1 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 7a671b996..da63b4d5e 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -342,6 +342,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;type=friend
;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
+ ; on incoming calls to Asterisk
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk