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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2009-01-28 13:07:27 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2009-01-28 13:07:27 +0000
commitdbd9e30d894a6b9399cb95e5aa90ead4a2802e11 (patch)
treef960c5ac69adf568483a600805de5d805626c84a /configs
parentd2a1102815120fa87677531f844ecea1962d3ca0 (diff)
Add a better explanation of the difference between the device namespace and the dialplan for newbies.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@171837 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rw-r--r--configs/sip.conf.sample19
1 files changed, 19 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 25bbb7735..c55f71bc6 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -22,6 +22,25 @@
; module reload chan_sip.so Reload configuration file
; Active SIP peers will not be reconfigured
;
+;------- Naming devices ------------------------------------------------------
+;
+; When naming devices, make sure you understand how Asterisk matches calls
+; that come in.
+; 1. Asterisk checks the SIP From: address username and matches against
+; names of devices with type=user
+; The name is the text between square brackets [name]
+; 2. Asterisk checks the IP address (and port number) that the INVITE
+; was sent from and matches against any devices with type=peer
+;
+; Don't mix extensions with the names of the devices. Devices need a unique
+; name. The device name is *not* used as phone numbers. Phone numbers are
+; anything you declare as an extension in the dialplan (extensions.conf).
+;
+; Note: The parameter "username" is not the username and in most cases is
+; not needed at all. Check below. In later releases, it's renamed
+; to "defaultuser" which is a better name, since it is used in
+; combination with the "defaultip" setting.
+;-----------------------------------------------------------------------------
[general]
context=default ; Default context for incoming calls