From: Joshua Colp Date: Thu, 30 Nov 2006 17:55:23 +0000 (+0000) Subject: Document 'port' for SIP peers, came up because of the current mailing list thread... X-Git-Tag: 1.2.14~30 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=37816992302511193a98726670990e708f134632;p=thirdparty%2Fasterisk.git Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage) 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 f080e149b4..b16eed5e7c 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