[asterisk-commits] rizzo: trunk r44579 - /trunk/configs/sip.conf.sample

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Fri Oct 6 09:43:36 MST 2006


Author: rizzo
Date: Fri Oct  6 11:43:36 2006
New Revision: 44579

URL: http://svn.digium.com/view/asterisk?rev=44579&view=rev
Log:
document a bit the use of templates.
They are highly convenient for writing configuration files, especially
if you have many similar entries, or want to switch quickly between
different configurations without having to comment/uncomment large
sections of the files.


Modified:
    trunk/configs/sip.conf.sample

Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?rev=44579&r1=44578&r2=44579&view=diff
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Fri Oct  6 11:43:36 2006
@@ -486,7 +486,51 @@
 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
 ; you will need to configure nat option for those phones.
 ; Also, turn on qualify=yes to keep the nat session open
-
+;
+; Because you might have a large number of similar sections, it is generally
+; convenient to use templates for the common parameters, and add them
+; the the various sections. Examples are below, and we can even leave
+; the templates uncommented as they will not harm:
+
+[basic-options](!)		; a template
+	dtmfmode=rfc2833
+	context=from-office
+	type=friend
+
+[natted-phone](!,basic-options)	; another template inheriting basic-options
+	nat=yes
+	canreinvite=no
+	host=dynamic
+
+[public-phone](!,basic-options)	; another template inheriting basic-options
+	nat=no
+	canreinvite=yes
+
+[my-codecs](!)		; a template for my preferred codecs
+	disallow=all
+	allow=ilbc
+	allow=g729
+	allow=gsm
+	allow=g723
+	allow=ulaw
+
+[ulaw-phone](!)		; and another one for ulaw-only
+	disallow=all
+	allow=ulaw
+
+; and finally instantiate a few phones
+;
+; [2133](natted-phone,my-codecs)
+;	secret = peekaboo
+; [2134](natted-phone,ulaw-hone)
+;	secret = not_very_secret
+; [2136](public-phone,ulaw-hone)
+;	secret = not_very_secret_either
+; ...
+;
+
+; Standard configurations not using templates look like this:
+;
 ;[grandstream1]
 ;type=friend 			
 ;context=from-sip		; Where to start in the dialplan when this phone calls



More information about the asterisk-commits mailing list