(( My ser.cfg is linked to at the end of the email. )) Basically, right now I have an AudioCodes FXO device (10.66.7.250) that registers with the proxy just fine, and when I dial out1 from my softphone (10.66.4.254), it dials to analog port 1 on the FXO, and out2 dials to analog port 2 on the FXO. I can do outbound dialing just fine. However, if I call the phone line plugged into either port, the requests hit the proxy (10.66.7.253) but don't ring the softphone. For example, here is the output from tethereal if I try calling in to port 2: 4260.972077 10.66.7.250 -> 10.66.7.253 SIP/SDP Request: INVITE sip:nbya2@nby-devproxy-01.tlsp.com, with session description 4260.972712 10.66.7.253 -> 10.66.7.250 SIP Status: 404 Not Found 4261.002006 10.66.7.250 -> 10.66.7.253 SIP Request: ACK sip:nbya2@nby-devproxy-01.tlsp.com My softphone (SJphone) is currently registered with the proxy with username nbya2: mysql> select username,domain,contact from location; +--------------+--------+------------------------------+ | username | domain | contact | +--------------+--------+------------------------------+ | nby-mp108-01 | | sip:nby-mp108-01@10.66.7.250 | | nbya1 | | sip:nbya1@10.66.5.99:5060 | | nbya2 | | sip:nbya2@10.66.4.254:5060 | +--------------+--------+------------------------------+ 3 rows in set (0.00 sec) Here is the relevant output from serctrl ul show for that user: ...Record(0x402cd1f8)... domain: 'location' aor : 'nbya2' ~~~Contact(0x402cd238)~~~ domain : 'location' aor : 'nbya2' Contact : 'sip:nbya2@10.66.4.254:5060' Expires : 2846 q : 0.00 Call-ID : '0D8E60D2-D1C3-439E-B099-3A327B6EB737(a)10.66.4.254' CSeq : 2 replic : 0 User-Agent: 'SJLabs-SJphone/1.40.258' State : CS_SYNC Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... I've tried dialing from both softphones that you can see listed there in the mysql output, but they don't contact each other either. Here is my ser config: http://aslan.homelinux.com/dana/tmp/ser.cfg So, I'm unsure where to go from here. This proxy is only on my LAN, if that makes any difference to you. I'd prefer to not change to stateful, unless it is mandatory for user lookup, but I didn't find any documentation yet that said that it was. Please let me know what I'm doing wrong. I've been using Google, the admin guide, examples from SIP.edu, voip-info.org, and I've tried the IRC chatroom, but it's quiet right now. -- Dana