mirror of
https://github.com/asterisk/asterisk.git
synced 2025-11-08 19:08:14 +00:00
Make it clear that caller ID in sip.conf is used only on incoming calls (inspired by bug #6183)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
@@ -342,6 +342,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||||||
;type=friend
|
;type=friend
|
||||||
;context=from-sip ; Where to start in the dialplan when this phone calls
|
;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
|
;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
|
;host=192.168.0.23 ; we have a static but private IP address
|
||||||
; No registration allowed
|
; No registration allowed
|
||||||
;nat=no ; there is not NAT between phone and Asterisk
|
;nat=no ; there is not NAT between phone and Asterisk
|
||||||
|
|||||||
Reference in New Issue
Block a user