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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2009-01-29 10:14:16 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2009-01-29 10:14:16 +0000
commit01e52ed0ae85f72627e84fdb2e6d77be04a749aa (patch)
treec314913b7f58ce30d860ef992574e9522cc33de9 /configs
parent856edee8fd7d1e3238cc741c9437eac25e133d06 (diff)
Merged revisions 171880 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r171880 | oej | 2009-01-28 14:26:31 +0100 (Ons, 28 Jan 2009) | 2 lines Add some more notes about device matching. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@172218 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rw-r--r--configs/sip.conf.sample22
1 files changed, 21 insertions, 1 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 901e99fce..6cb294581 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -58,12 +58,18 @@
; 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
+; 2. Asterisk checks the From: addres and matches the list of devices
+; with a type=peer
+; 3. 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).
+;
+; When setting up trunks, make sure there's no risk that any From: username
+; (caller ID) will match any of your device names, because then Asterisk
+; might match the wrong device.
;
; 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
@@ -446,6 +452,20 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; and more readable because you don't have to write the parameters in two places
; (note that the "port" is ignored - this is a bug that should be fixed).
;
+; Note that a register= line doesn't mean that we will match the incoming call in any
+; other way than described above. If you want to control where the call enters your
+; dialplan, which context, you want to define a peer with the hostname of the provider's
+; server. If the provider has multiple servers to place calls to your system, you need
+; a peer for each server.
+;
+; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
+; contain a port number. Since the logical separator between a host and port number is a
+; ':' character, and this character is already used to separate between the optional "secret"
+; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
+; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
+; they are blank. See the third example below for an illustration.
+;
+;
; Examples:
;
;register => 1234:password@mysipprovider.com