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Diffstat (limited to 'configs/sip.conf.sample')
-rw-r--r-- | configs/sip.conf.sample | 19 |
1 files changed, 19 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 513d45418..901e99fce 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 |