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-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 5fd2657ba..bde424ba9 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -51,6 +51,25 @@
; 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.
+;-----------------------------------------------------------------------------
; ** Deprecated configuration options **
; The "call-limit" configuation option is deprecated. It still works in