[Freeswitch-users] Wrong IP on ACK?

[David Aldworth]

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 at 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 at 64.74.188.23>;tag=Dt6v81cDZXa3B.
To: <sip:989607XXXX at 68.188.189.202:5060>.
Call-ID: de7c471c-252e-122c-3cba-5f1bac93b621.
CSeq: 106789378 INVITE.
Contact: <sip:mod_sofia at 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 at 64.74.188.23>;tag=4UF788r8ct8aD.
To: <sip:989607XXXX at 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 at 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 at 64.74.188.23>;tag=4UF788r8ct8aD.
To: <sip:989607XXXX at 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 at 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 at 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 at 64.74.188.23>;tag=4UF788r8ct8aD.
To: <sip:989607XXXX at 68.188.189.202:5060>;tag=as1da4b7aa.
Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
CSeq: 106789510 ACK.
Contact: <sip:mod_sofia at 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>