For some reason when trunking with Asterisk PBX's (yes, I know) FS wants to send the ACK to the internal ip found in the Contact field of the 200 OK. We have the force rport setting on but it's still not responding to that IP. Register's work. Most of the sip signalling works, just when the customer specifies the Contact filed with an internal ip. Below is a packet capture and our external.xml conf file. U 2008/11/04 09:17:08.259672 64.74.188.23:5060 -> 68.188.189.202:5060 INVITE sip:989607XXXX@68.188.189.202:5060 SIP/2.0. Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bKc6y6pr80HyeaN. Max-Forwards: 68. From: "user" <sip:303452XXXX@64.74.188.23>;tag=Dt6v81cDZXa3B. To: <sip:989607XXXX@68.188.189.202:5060>. Call-ID: de7c471c-252e-122c-3cba-5f1bac93b621. CSeq: 106789378 INVITE. Contact: <sip:mod_sofia@64.74.188.23:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session- description, presence.winfo, message-summary. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 370. U 2008/11/04 09:21:32.861720 68.188.189.202:5060 -> 64.74.188.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 64.74.188.23 ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:989607XXXX@192.168.0.5>. Content-Length: 0. . U 2008/11/04 09:21:32.861845 68.188.189.202:5060 -> 64.74.188.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 64.74.188.23 ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:989607XXXX@192.168.0.5>. Content-Type: application/sdp. Content-Length: 285. . v=0. o=root 10970 10970 IN IP4 192.168.0.5. s=session. c=IN IP4 192.168.0.5. t=0 0. m=audio 15876 RTP/AVP 18 0 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2008/11/04 09:21:32.862573 64.74.188.23:5060 -> 192.168.0.5:5060 ACK sip:989607XXXX@192.168.0.5 SIP/2.0. Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bK0r7c2a501SQvD. Max-Forwards: 70. From: "user" <sip:303452XXXX@64.74.188.23>;tag=4UF788r8ct8aD. To: <sip:989607XXXX@68.188.189.202:5060>;tag=as1da4b7aa. Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621. CSeq: 106789510 ACK. Contact: <sip:mod_sofia@64.74.188.23:5060>. Content-Length: 0. . external.xml <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings>