Hi there, Ive been trying to get my first elastix box setup for the last two weeks and after hundreds of google searches, Im out of ideas and in search of help. Thanks in advance for any assistance that can be provided. Elastix Version: 1.3-2 Asterisk: Asterisk 1.4.22-rc5 LAN setp: Linksys wrt310n router, Elastix box has been placed in the DMZ, firewall on the router has been turned off Everything works great, but I cant seem to get both inbound and outbound calling working at the same time over my voipvoip.com SIP trunk. It seems like a NAT issue, because when I adjust the nat setting in the trunk confguration, it will fix the problem I have with inbound, for instance, but break the outbound. Nothing I have found yet has solved both problems at once. At the moment, I have set my trunk configuration back to the exact reccomended settings from VoipVoip.com's config guide: Code: Outbound Caller ID: *My 10 digit DID number* Dial Rules: 1415+NXXXXXX (For 7 digit dialing) Outgoing Settings Trunk Name: voipvoip PEER Details: username=*my SIP number for voipvoip* type=peer secret=****** nat=auto insecure=very host=sip3.voipvoip.com fromuser=*My SIP Number* fromdomain=sip3.voipvoip.com dtmfmode=rfc2833 disallow=all allow=g729&ilbc&ulaw&alaw Incoming Settings: User Context: *My SIP Number* User Details: username=*My Sip Number from voipvoip* type=user secret=******** nat=auto insecure=very host=sip3.voipvoip.com fromdomain=sip3.voipvoip.com dtmfmode=rfc2833 disallow=all allow=g729&ulaw&alaw&ilbc context=from-trunk Registration String: My SIP Number:*******@sip3.voipvoip.com/My Sip Number With this setup, I can make calls most of the time. When I try to dial in from my cell phone, the call is just silent, no ringing, no nothing. Sometimes, its an error from voipvoip.com saying that my extension is not available, as if Im not logged in. When the outbound calls stop working, inbound calls will work. When inbound calls are working and I do a "sip show registry" I see this: Code: Host Username Refresh State Reg.Time sip3.voipvoip.com:5060 *My SIP Number* 105 Registered Sun, 01 Feb 2009 12:45:09 When inbound calls are working and I do a "sip show channels" I see this: Code: Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 220.127.116.11 (None) e720f306-f0 00101/00001 0x0 (nothing) No Rx: OPTIONS 18.104.22.168 (None) 9b8ab9a3-0b 00101/00001 0x0 (nothing) No Rx: OPTIONS 2 active SIP channels When inbound calls stop working, but outbound calls are working, here are the results: "sip show registry" Code: Host Username Refresh State Reg.Time sip3.voipvoip.com:5060 *My Sip Number" 105 Registered Sun, 01 Feb 2009 12:59:10 "sip show channels" Code: Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 22.214.171.124 (None) 9b8ab9a3-e5 00101/00001 0x0 (nothing) No Rx: OPTIONS 1 active SIP channel When I add the following code to sip_custom.conf, I get different results: Code: nat=yes externhost=<My IP> externrefresh=10 localnet=126.96.36.199/255.255.255.0 bindport = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) With that code added, I receive calls perfectly, but get an "all circuits are busy" error when I call outbound. Can anyone help me out? Thanks for taking the time to read this.