From: Joshua Colp Date: Thu, 30 Nov 2006 17:57:35 +0000 (+0000) Subject: Merged revisions 48142 via svnmerge from X-Git-Tag: 1.4.0-beta4~57 X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=802c3c3ecff4647040c8591e59d295b622326048;p=thirdparty%2Fasterisk.git Merged revisions 48142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48143 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 35e9dffad9..52c65cc0d1 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -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