From 0916c53ea0a9fbb7d4e4a6091f6d175e9b9a7514 Mon Sep 17 00:00:00 2001 From: file Date: Thu, 30 Nov 2006 17:55:23 +0000 Subject: 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 --- configs/sip.conf.sample | 2 ++ 1 file changed, 2 insertions(+) (limited to 'configs/sip.conf.sample') 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 -- cgit v1.2.3