Merged revisions 48143 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

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r48143 | file | 2006-11-30 12:57:35 -0500 (Thu, 30 Nov 2006) | 10 lines

Merged revisions 48142 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines

Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48144 65c4cc65-6c06-0410-ace0-fbb531ad65f3

configs/sip.conf.sample

@@ -39,6 +39,7 @@ allowoverlap=no			; Disable overlap dialing support. (Default is yes)
 				; 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 
@@ -500,8 +501,9 @@ 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
-				; Call-limits will not be enforced on real-time peers,
-				; since they are not stored in-memory
+					; Call-limits will not be enforced on real-time peers,
+					; since they are not stored in-memory
+;port=80				; The port number we want to connect to on the remote side
 
 ;--- sample definition for a provider
 ;[provider1]