[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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