Hello I'm having a difficult time finding precisely what to put in sip.conf and extensions.conf (and possibly other files) to get a working configuration to connect an Asterisk (1.4) server to a VoIP provider with the Asterisk server + SIP clients located in a private LAN behind a NAT router:
http://img560.imageshack.us/img560/3749/asterisknat.png Would someone have a full, direct (ie. doesn't depend on GUIs like FreePBX, etc.) working example that I could look at as reference? Thank you. PS: Here's what I'm thinking of using: ;====================== sip.conf [general] ;map this UDP port on NAT router port = 5060 bindaddr = 0.0.0.0 ;just to be safe context = dummy deny=0.0.0.0/0 permit=<IP address of VOSP server> externip=<public IP address of NAT router> localnet=192.168.0.0/24 disallow=all allow=ulaw allow=alaw allow=gsm ;all RTP packets go through Asterisk canreinvite=no ;incoming calls from VOSP register => me:mypas...@mysipprovider.com ;for outgoing calls to VOSP [vosp] ;friend = peer+user type=friend username=me fromuser=me fromdomain=mysipprovider.com authname=me secret=mypasswd host=mysipprovider.com insecure=very qualify=yes context=outgoing ;Since VOSP is on the Net, nat=no or nat=yes? nat=no ;extension for XLite [6011] type=friend context=internal secret=6011 host=dynamic ;client on same LAN as Asterisk nat=no ;extension for IP phone [6012] type=friend context=internal secret=6012 host=dynamic ;client on same LAN as Asterisk nat=no ;====================== extensions.conf [general] static=yes writeprotect=yes clearglobalvars=no autofallthrough=yes [vosp-incoming] exten => s,1,Dial(SIP/6011) exten => s,n,Hangup [internal] exten => 6011,1,Dial(SIP/6011) exten => 6011,n,Hangup exten => 6012,1,Dial(SIP/6012) exten => 6012,n,Hangup include => outgoing [outgoing] ;Route calls starting with 0 to VOSP exten => _0.,1,Dial(SIP/vosp/${EXTEN}) exten => _0.,n,Hangup ;====================== rtp.conf [general] rtpstart=10000 ;1 even port for (symetric) RTP + 1 odd port for RTCP ;for a total of 10 concurrent conversations rtpend=10020 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users