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:[EMAIL PROTECTED]:5060 SIP/2.0.
Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bKc6y6pr80HyeaN.
Max-Forwards: 68.
From: "user" <sip:[EMAIL PROTECTED]>;tag=Dt6v81cDZXa3B.
To: <sip:[EMAIL PROTECTED]:5060>.
Call-ID: de7c471c-252e-122c-3cba-5f1bac93b621.
CSeq: 106789378 INVITE.
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=4UF788r8ct8aD.
To: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>.
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:[EMAIL PROTECTED]>;tag=4UF788r8ct8aD.
To: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>.
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:[EMAIL PROTECTED] SIP/2.0.
Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bK0r7c2a501SQvD.
Max-Forwards: 70.
From: "user" <sip:[EMAIL PROTECTED]>;tag=4UF788r8ct8aD.
To: <sip:[EMAIL PROTECTED]:5060>;tag=as1da4b7aa.
Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
CSeq: 106789510 ACK.
Contact: <sip:[EMAIL PROTECTED]: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>
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