[asterisk-commits] mmichelson: trunk r190577 - /trunk/configs/sip.conf.sample
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Apr 27 09:46:17 CDT 2009
Author: mmichelson
Date: Mon Apr 27 09:46:14 2009
New Revision: 190577
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=190577
Log:
Remove nonexistent option from sip.conf.sample.
The option to choose which connected line header to
use is not 'rpid_header' but 'sendrpid'
Modified:
trunk/configs/sip.conf.sample
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/svn-view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=190577&r1=190576&r2=190577
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Mon Apr 27 09:46:14 2009
@@ -228,9 +228,6 @@
; This is identical to sendrpid=yes
;sendrpid = pai ; Use the "P-Asserted-Identity" header
; to send the identity of the remote party
-;rpid_header = rpid ; Which header should be used when sending Remote Party ID
- ; 'rpid' means to send "Remote-Party-ID"
- ; 'pai' means to send "P-Asserted-Identity"
;progressinband=never ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it
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