<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">We're still having a problem with this. As you can see from the below the ACK goes to the port in the Contact field of the 200 OK instead of that of the UDP header, which is where their router is expecting to get the call from.<div><br></div><div>Help!</div><div><br></div><div>David</div><div><br><div><div>On Nov 6, 2008, at 10:55 AM, David Aldworth wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">No love. They set extern ip so the IP comes through correctly, but the acl did not seem to have any affect. We are still sending to the wrong port. Sip trace, acl.conf.xml and sip profile are below:<div><br></div><div><div>U 2008/11/06 10:46:01.924795 70.88.65.1:50085 -> 70.42.223.23:5060</div><div>SIP/2.0 100 Trying.</div><div>Via: SIP/2.0/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.</div><div>From: "TELIAX FAX" <<a href="sip:303825XXXX@70.42.223.23">sip:303825XXXX@70.42.223.23</a>>;tag=armgX7QeNQ94N.</div><div>To: <<a href="sip:317376XXXX@70.88.65.1:50085">sip:317376XXXX@70.88.65.1:50085</a>>.</div><div>Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.</div><div>CSeq: 106878444 INVITE.</div><div>User-Agent: Asterisk PBX.</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.</div><div>Contact: <<a href="sip:317376XXXX@70.88.65.1">sip:317376XXXX@70.88.65.1</a>>.</div><div>Content-Length: 0.</div><div>.</div><div><br></div><div><div>U 2008/11/06 10:46:01.931791 70.88.65.1:50085 -> 70.42.223.23:5060</div><div>SIP/2.0 180 Ringing.</div><div>Via: SIP/2.0/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.</div><div>From: "TELIAX FAX" <<a href="sip:303825XXXX@70.42.223.23">sip:303825XXXX@70.42.223.23</a>>;tag=armgX7QeNQ94N.</div><div>To: <<a href="sip:317376XXXX@70.88.65.1:50085">sip:317376XXXX@70.88.65.1:50085</a>>;tag=as78a21a0c.</div><div>Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.</div><div>CSeq: 106878444 INVITE.</div><div>User-Agent: Asterisk PBX.</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.</div><div>Contact: <<a href="sip:317376XXXX@70.88.65.1">sip:317376XXXX@70.88.65.1</a>>.</div><div>Content-Length: 0.</div><div>.</div><div><br></div><div><div>U 2008/11/06 10:46:01.932294 70.88.65.1:50085 -> 70.42.223.23:5060</div><div>SIP/2.0 200 OK.</div><div>Via: SIP/2.0/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.</div><div>From: "TELIAX FAX" <<a href="sip:303825XXXX@70.42.223.23">sip:303825XXXX@70.42.223.23</a>>;tag=armgX7QeNQ94N.</div><div>To: <<a href="sip:317376XXXX@70.88.65.1:50085">sip:317376XXXX@70.88.65.1:50085</a>>;tag=as78a21a0c.</div><div>Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.</div><div>CSeq: 106878444 INVITE.</div><div>User-Agent: Asterisk PBX.</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.</div><div>Contact: <<a href="sip:317376XXXX@70.88.65.1">sip:317376XXXX@70.88.65.1</a>>.</div><div>Content-Type: application/sdp.</div><div>Content-Length: 257.</div><div>.</div><div>v=0.</div><div>o=root 2901 2901 IN IP4 70.88.65.1.</div><div>s=session.</div><div>c=IN IP4 70.88.65.1.</div><div>t=0 0.</div><div>m=audio 19378 RTP/AVP 0 8 3 101.</div><div>a=rtpmap:0 PCMU/8000.</div><div>a=rtpmap:8 PCMA/8000.</div><div>a=rtpmap:3 GSM/8000.</div><div>a=rtpmap:101 telephone-event/8000.</div><div>a=fmtp:101 0-16.</div><div>a=silenceSupp:off - - - -.</div><div><br></div><div><div>U 2008/11/06 10:46:01.932694 70.42.223.23:5060 -> 70.88.65.1:5060</div><div>ACK <a href="sip:317376XXXX@70.88.65.1">sip:317376XXXX@70.88.65.1</a> SIP/2.0.</div><div>Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc.</div><div>Max-Forwards: 70.</div><div>From: "TELIAX FAX" <<a href="sip:303825XXXX@70.42.223.23">sip:303825XXXX@70.42.223.23</a>>;tag=armgX7QeNQ94N.</div><div>To: <<a href="sip:317376XXXX@70.88.65.1:50085">sip:317376XXXX@70.88.65.1:50085</a>>;tag=as78a21a0c.</div><div>Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.</div><div>CSeq: 106878444 ACK.</div><div>Contact: <<a href="sip:mod_sofia@70.42.223.23:5060">sip:mod_sofia@70.42.223.23:5060</a>>.</div><div>Content-Length: 0.</div><div><br></div><div><br></div><div>Here is the acl:</div><div><br></div><div><div><configuration name="acl.conf" description="Network Lists"></div><div> <network-lists></div><div> <list name="dl-candidates" default="allow"></div><div> <node type="deny" cidr="10.0.0.0/8"/></div><div> <node type="deny" cidr="172.16.0.0/12"/></div><div> <node type="deny" cidr="192.168.0.0/16"/></div><div> </list></div><div> <list name="rfc1918" default="deny"></div><div> <node type="allow" cidr="10.0.0.0/8"/></div><div> <node type="allow" cidr="172.16.0.0/12"/></div><div> <node type="allow" cidr="192.168.0.0/16"/></div><div> </list></div><div> <list name="lan" default="allow"></div><div> <node type="deny" cidr="192.168.42.0/24"/></div><div> <node type="allow" cidr="192.168.42.42/32"/></div><div> </list></div><div> <list name="strict" default="deny"></div><div> <node type="allow" cidr="208.102.123.124/32"/></div><div> </list></div><div> <list name="domains" default="deny"></div><div> <node type="allow" domain="$${domain}"/></div><div> </list></div><div> <list name="nat" default="allow"></div><div> <node type="allow" cidr="0.0.0.0/0"/></div><div> </list></div><div> </network-lists></div><div></configuration></div><div><br></div><div><br></div><div>And here is the sip profile:</div><div><br></div><div><div><profile name="external"></div><div><br></div><div> <gateways></div><div> <X-PRE-PROCESS cmd="include" data="external/*.xml"/></div><div> </gateways></div><div> </div><div> <domains></div><div> <domain name="$${domain}" parse="true"/></div><div> </domains></div><div><br></div><div> <settings></div><div> <param name="debug" value="0"/></div><div> <param name="sip-trace" value="no"/></div><div> <param name="rfc2833-pt" value="101"/></div><div> <param name="sip-port" value="5060"/></div><div> <param name="dialplan" value="XML"/></div><div> <param name="context" value="public"/></div><div> <param name="dtmf-duration" value="100"/></div><div> <param name="codec-prefs" value="$${outbound_codec_prefs}"/></div><div> <param name="hold-music" value="$${hold_music}"/></div><div> <param name="use-rtp-timer" value="true"/></div><div> <param name="rtp-timer-name" value="soft"/></div><div> <param name="multiple-registrations" value="true"/></div><div> <param name="manage-presence" value="true"/></div><div> <param name="aggressive-nat-detection" value="true"/></div><div> <param name="NDLB-force-rport" value="true"/></div><div> <param name="inbound-codec-negotiation" value="generous"/></div><div> <param name="nonce-ttl" value="60"/></div><div> <param name="auth-calls" value="true"/></div><div> <param name="rtp-timeout-sec" value="1800"/></div><div> <param name="rtp-ip" value="$${local_ip_v4}"/></div><div> <param name="sip-ip" value="$${local_ip_v4}"/></div><div> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/></div><div> <param name="ext-sip-ip" value="$${external_sip_ip}"/></div><div> <param name="rtp-timeout-sec" value="300"/></div><div> <param name="rtp-hold-timeout-sec" value="1800"/></div><div> <param name="apply-nat-acl" value="nat"/></div><div> </settings></div><div></profile></div></div></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div></div></div></div><div><div>On Nov 6, 2008, at 8:37 AM, Anthony Minessale wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">doh,<br>I keep doing that sorry.<br><br>apply-nat-acl not apply_nat_acl <br><br><br><br><div class="gmail_quote">On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <span dir="ltr"><<a href="mailto:daldworth@teliax.com">daldworth@teliax.com</a>></span> wrote:<br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div style="">Yes. Below are settings that have been persistent through recent testing. Is there anything else we can try or should we open a jira?<div> <br></div><div><div><div></div><div class="Wj3C7c"><div> <settings></div><div> <param name="debug" value="0"/></div><div> <param name="sip-trace" value="no"/></div> <div> <param name="rfc2833-pt" value="101"/></div><div> <param name="sip-port" value="5060"/></div><div> <param name="dialplan" value="XML"/></div> <div> <param name="context" value="public"/></div><div> <param name="dtmf-duration" value="100"/></div><div> <param name="codec-prefs" value="$${outbound_codec_prefs}"/></div> <div> <param name="hold-music" value="$${hold_music}"/></div><div> <param name="use-rtp-timer" value="true"/></div><div> <param name="rtp-timer-name" value="soft"/></div> <div> <param name="multiple-registrations" value="true"/></div><div> <param name="manage-presence" value="true"/></div><div> <param name="aggressive-nat-detection" value="true"/></div> <div> <param name="NDLB-force-rport" value="true"/></div><div> <param name="inbound-codec-negotiation" value="generous"/></div><div> <param name="nonce-ttl" value="60"/></div> <div> <param name="auth-calls" value="true"/></div><div> <param name="rtp-timeout-sec" value="1800"/></div><div> <param name="rtp-ip" value="$${local_ip_v4}"/></div> <div> <param name="sip-ip" value="$${local_ip_v4}"/></div><div> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/></div><div> <param name="ext-sip-ip" value="$${external_sip_ip}"/></div> <div> <param name="rtp-timeout-sec" value="300"/></div><div> <param name="rtp-hold-timeout-sec" value="1800"/></div></div></div><div class="Ih2E3d"><div> <param name="apply_nat_acl" value="nat"/></div> </div><div> </settings></div><div><div></div><div class="Wj3C7c"><div><br></div><div><div>On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:</div><br><blockquote type="cite">did you remember to add<br><param name="apply_nat_acl" value="nat"/><br> to the profile in question and restart?<br><br><div class="gmail_quote">On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <span dir="ltr"><<a href="mailto:daldworth@teliax.com" target="_blank">daldworth@teliax.com</a>></span> wrote:<br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Brian, we updated the acl to:<br> <div><br> <list name="nat" default="allow"><br> </div> <node type="allow" cidr="<a href="http://0.0.0.0/0" target="_blank">0.0.0.0/0</a>"/><br> </list><br> <br> And the ACK is still going to the wrong (right but wrong) ip/port.<br> <br> Is there any way to get that ACK to go to the ip/port of the UDP header?<br> <font color="#888888"><br> David<br> </font><div><div></div><div><br> On Nov 5, 2008, at 4:21 PM, Brian West wrote:<br> <br> > <a href="http://0.0.0.0/0" target="_blank">0.0.0.0/0</a> should match all IP space.<br> ><br> > /b<br> ><br> > On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:<br> ><br> >> Anthony, In hopes of matching all IP's we added a very simple:<br> >><br> >> <list name="nat" default="allow"><br> >> </list><br> >><br> >> To the acl.conf.xml and we added:<br> >><br> >> <param name="apply_nat_acl" value="nat"/><br> >><br> >> To the sip profile. Unfortunately there was no affect. What would be<br> >> the correct acl to match all IP's?<br> >><br> >> David<br> ><br> ><br> > _______________________________________________<br> > Freeswitch-users mailing list<br> > <a href="mailto:Freeswitch-users@lists.freeswitch.org" target="_blank">Freeswitch-users@lists.freeswitch.org</a><br> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br> > UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br> <br> <br> _______________________________________________<br> Freeswitch-users mailing list<br> <a href="mailto:Freeswitch-users@lists.freeswitch.org" target="_blank">Freeswitch-users@lists.freeswitch.org</a><br> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br> </div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br> ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br> <br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a><br> IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br> <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br> <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org" target="_blank">googletalk:conf+888@conference.freeswitch.org</a><br> pstn:213-799-1400<br> _______________________________________________<br>Freeswitch-users mailing list<br><a href="mailto:Freeswitch-users@lists.freeswitch.org" target="_blank">Freeswitch-users@lists.freeswitch.org</a><br> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br></div></div></div></div><br>_______________________________________________<br> Freeswitch-users mailing list<br> <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br> <br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br> <br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br> IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br> <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400<br> _______________________________________________<br>Freeswitch-users mailing list<br><a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br><a 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