Document 'port' for SIP peers, came up because of the current mailing list thread...

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@48142 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index f080e149b468661aaeaf930507f6a963106fda73..b16eed5e7c625801d33bf84e33ab38b04db9f3e0 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -32,6 +32,7 @@ context=default                       ; Default context for incoming calls
                                ; 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 is 5060)
+                               ; bindport is the local UDP port that Asterisk will listen on
 bindaddr=0.0.0.0               ; IP address to bind to (0.0.0.0 binds to all)
 srvlookup=yes                  ; Enable DNS SRV lookups on outbound calls
                                ; Note: Asterisk only uses the first host 
@@ -327,6 +328,7 @@ srvlookup=yes                       ; Enable DNS SRV lookups on outbound calls
 ;usereqphone=yes                       ; This provider requires ";user=phone" on URI
 ;call-limit=5                          ; permit only 5 simultaneous outgoing calls to this peer
 ;outboundproxy=proxy.provider.domain   ; send outbound signaling to this proxy, not directly to the peer
+;port=80                               ; The port number we want to connect to on the remote side
 
 ;------------------------------------------------------------------------------
 ; Definitions of locally connected SIP phones