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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2008-07-06 20:19:04 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2008-07-06 20:19:04 +0000
commitccb307b7a87bd930f286efcf887513f034cab9da (patch)
tree927069ddda2f4c3c90139354d168893fa6c8813f /configs
parent1420f15197af64e05b5648c893df70c33130ee36 (diff)
- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" and "tlsbindaddr".
Note: I don't think we can start properly without UDP port open, that needs to be tested. - Removing "bindport" from configuration example, not needed to mention this any more I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128525 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rw-r--r--configs/sip.conf.sample7
1 files changed, 2 insertions, 5 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 64adbf2c1..54f497328 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -75,11 +75,8 @@ allowoverlap=no ; Disable overlap dialing support. (Default is yes)
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
-bindport=5060 ; UDP Port to bind to (SIP standard port for unencrypted UDP
- ; and TCP sessions is 5060)
- ; bindport is the local UDP port that Asterisk will listen on
-bindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
- ; You can specify port here too, like 123.123.123.123:5080
+udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
+ ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
;
; Note that the TCP and TLS support for chan_sip is currently considered