[Asterisk-cvs] asterisk/configs sip.conf.sample,1.70,1.71

[kpfleming]

Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv24487/configs

Modified Files:
	sip.conf.sample 
Log Message:
support optional sending of Remote-Party-ID headers (issue #2471, heavily modified to actually work properly)


Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.70
retrieving revision 1.71
diff -u -d -r1.70 -r1.71
--- sip.conf.sample	26 Sep 2005 23:14:59 -0000	1.70
+++ sip.conf.sample	27 Sep 2005 01:54:17 -0000	1.71
@@ -69,6 +69,7 @@
 ;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP activity
 				; when we're on hold (must be > rtptimeout)
 ;trustrpid = no			; If Remote-Party-ID should be trusted
+;sendrpid = yes			; If Remote-Party-ID should be sent
 ;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
@@ -267,6 +268,7 @@
 ;                             defaultip
 ;                             rtptimeout
 ;                             rtpholdtimeout
+;                             sendrpid
 
 ;[sip_proxy]
 ; For incoming calls only. Example: FWD (Free World Dialup)