Re: [Freeswitch-users] Wrong IP on ACK?

David Aldworth Mon, 10 Nov 2008 15:35:35 -0800

Anthony. Did you want to log in and check it out?

I can send you the files if you think it's something else.


David

On Nov 6, 2008, at 12:38 PM, Anthony Minessale wrote:

This is svn trunk? There is no reason this should not work. ithappens all the time where this setting breaks it for people goingthe other way when they don't want it to happen.


If you can't get it working we can probably configure it for you.

On Thu, Nov 6, 2008 at 11:55 AM, David Aldworth<[EMAIL PROTECTED]> wrote:No love. They set extern ip so the IP comes through correctly, butthe acl did not seem to have any affect. We are still sending to thewrong port. Sip trace, acl.conf.xml and sip profile are below:


U 2008/11/06 10:46:01.924795 70.88.65.1:50085 -> 70.42.223.23:5060
SIP/2.0 100 Trying.

Via: SIP/2.0/UDP70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.

From: "TELIAX FAX" <sip:[EMAIL PROTECTED]>;tag=armgX7QeNQ94N.
To: <sip:[EMAIL PROTECTED]:50085>.
Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.
CSeq: 106878444 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <sip:[EMAIL PROTECTED]>.
Content-Length: 0.
.

U 2008/11/06 10:46:01.931791 70.88.65.1:50085 -> 70.42.223.23:5060
SIP/2.0 180 Ringing.

Via: SIP/2.0/UDP70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.

From: "TELIAX FAX" <sip:[EMAIL PROTECTED]>;tag=armgX7QeNQ94N.
To: <sip:[EMAIL PROTECTED]:50085>;tag=as78a21a0c.
Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.
CSeq: 106878444 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <sip:[EMAIL PROTECTED]>.
Content-Length: 0.
.

U 2008/11/06 10:46:01.932294 70.88.65.1:50085 -> 70.42.223.23:5060
SIP/2.0 200 OK.

Via: SIP/2.0/UDP70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.

From: "TELIAX FAX" <sip:[EMAIL PROTECTED]>;tag=armgX7QeNQ94N.
To: <sip:[EMAIL PROTECTED]:50085>;tag=as78a21a0c.
Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.
CSeq: 106878444 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <sip:[EMAIL PROTECTED]>.
Content-Type: application/sdp.
Content-Length: 257.
.
v=0.
o=root 2901 2901 IN IP4 70.88.65.1.
s=session.
c=IN IP4 70.88.65.1.
t=0 0.
m=audio 19378 RTP/AVP 0 8 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.

U 2008/11/06 10:46:01.932694 70.42.223.23:5060 -> 70.88.65.1:5060
ACK sip:[EMAIL PROTECTED] SIP/2.0.
Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc.
Max-Forwards: 70.
From: "TELIAX FAX" <sip:[EMAIL PROTECTED]>;tag=armgX7QeNQ94N.
To: <sip:[EMAIL PROTECTED]:50085>;tag=as78a21a0c.
Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.
CSeq: 106878444 ACK.
Contact: <sip:[EMAIL PROTECTED]:5060>.
Content-Length: 0.


Here is the acl:

<configuration name="acl.conf" description="Network Lists">
  <network-lists>
    <list name="dl-candidates" default="allow">
      <node type="deny" cidr="10.0.0.0/8"/>
      <node type="deny" cidr="172.16.0.0/12"/>
      <node type="deny" cidr="192.168.0.0/16"/>
    </list>
    <list name="rfc1918" default="deny">
      <node type="allow" cidr="10.0.0.0/8"/>
      <node type="allow" cidr="172.16.0.0/12"/>
      <node type="allow" cidr="192.168.0.0/16"/>
    </list>
    <list name="lan" default="allow">
      <node type="deny" cidr="192.168.42.0/24"/>
      <node type="allow" cidr="192.168.42.42/32"/>
    </list>
    <list name="strict" default="deny">
      <node type="allow" cidr="208.102.123.124/32"/>
    </list>
    <list name="domains" default="deny">
      <node type="allow" domain="$${domain}"/>
    </list>
    <list name="nat" default="allow">
      <node type="allow" cidr="0.0.0.0/0"/>
    </list>
  </network-lists>
</configuration>


And here is the sip profile:

<profile name="external">

  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <domains>
    <domain name="$${domain}" parse="true"/>
  </domains>

  <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"/>
    <param name="apply-nat-acl" value="nat"/>
  </settings>
</profile>






On Nov 6, 2008, at 8:37 AM, Anthony Minessale wrote:

doh,
I keep doing that sorry.

apply-nat-acl not apply_nat_acl

On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth<[EMAIL PROTECTED]> wrote:Yes. Below are settings that have been persistent through recenttesting. Is there anything else we can try or should we open a jira?


  <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"/>
    <param name="apply_nat_acl" value="nat"/>
  </settings>

On Nov 6, 2008, at 7:01 AM, Anthony Minessale wrote:

did you remember to add
<param name="apply_nat_acl" value="nat"/>
to the profile in question and restart?

On Wed, Nov 5, 2008 at 10:39 PM, David Aldworth <[EMAIL PROTECTED]> wrote:

Brian, we updated the acl to:

    <list name="nat" default="allow">
      <node type="allow" cidr="0.0.0.0/0"/>
    </list>

And the ACK is still going to the wrong (right but wrong) ip/port.

Is there any way to get that ACK to go to the ip/port of the UDPheader?


David

On Nov 5, 2008, at 4:21 PM, Brian West wrote:

> 0.0.0.0/0 should match all IP space.
>
> /b
>
> On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:
>
>> Anthony, In hopes of matching all IP's we added a very simple:
>>
>>    <list name="nat" default="allow">
>>    </list>
>>
>> To the acl.conf.xml and we added:
>>
>>    <param name="apply_nat_acl" value="nat"/>
>>

>> To the sip profile. Unfortunately there was no affect. Whatwould be

>> the correct acl to match all IP's?
>>
>> David
>
>
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--
Anthony Minessale II

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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:[EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED]
iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED]
pstn:213-799-1400
_______________________________________________
Freeswitch-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:[EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED]
iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED]
pstn:213-799-1400
_______________________________________________
Freeswitch-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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Re: [Freeswitch-users] Wrong IP on ACK?

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