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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-11-30 17:55:23 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-11-30 17:55:23 +0000
commit0916c53ea0a9fbb7d4e4a6091f6d175e9b9a7514 (patch)
tree8e1c1922d9dc4854bb00ccc8c6a152c0f9211592 /configs
parenteb38770dbe60e2496657184344eae6f9c7c34fda (diff)
Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2@48142 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rw-r--r--configs/sip.conf.sample2
1 files changed, 2 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index f080e149b..b16eed5e7 100644
--- 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