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Merged revisions 48142 via svnmerge from

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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/branches/1.4@48143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2006-11-30 17:57:35 +00:00
parent e49855c2bd
commit 802c3c3ecf
1 changed files with 4 additions and 2 deletions
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6
configs/sip.conf.sample
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@@ -35,6 +35,7 @@ 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
@@ -481,8 +482,9 @@ 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
;------------------------------------------------------------------------------
; Definitions of locally connected SIP devices