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[Freeswitch-users] Wrong IP on ACK?


 
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daldworth at teliax.com
Guest
PostPosted: Wed Nov 19, 2008 9:08 am    Post subject: [Freeswitch-users] Wrong IP on ACK? Reply with quote
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.

Help!


David

On Nov 6, 2008, at 10:55 AM, David Aldworth wrote:
Quote:
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:

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/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.
From: "TELIAX FAX" <[url=sip:303825XXXX@70.42.223.23]sip:303825XXXX@70.42.223.23[/url]>;tag=armgX7QeNQ94N.
To: <[url=sip:317376XXXX@70.88.65.1:50085]sip:317376XXXX@70.88.65.1:50085[/url]>.
Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.
CSeq: 106878444 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <[url=sip:317376XXXX@70.88.65.1]sip:317376XXXX@70.88.65.1[/url]>.
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/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.
From: "TELIAX FAX" <[url=sip:303825XXXX@70.42.223.23]sip:303825XXXX@70.42.223.23[/url]>;tag=armgX7QeNQ94N.
To: <[url=sip:317376XXXX@70.88.65.1:50085]sip:317376XXXX@70.88.65.1:50085[/url]>;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: <[url=sip:317376XXXX@70.88.65.1]sip:317376XXXX@70.88.65.1[/url]>.
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/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.
From: "TELIAX FAX" <[url=sip:303825XXXX@70.42.223.23]sip:303825XXXX@70.42.223.23[/url]>;tag=armgX7QeNQ94N.
To: <[url=sip:317376XXXX@70.88.65.1:50085]sip:317376XXXX@70.88.65.1:50085[/url]>;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: <[url=sip:317376XXXX@70.88.65.1]sip:317376XXXX@70.88.65.1[/url]>.
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 [url=sip:317376XXXX@70.88.65.1]sip:317376XXXX@70.88.65.1[/url] SIP/2.0.
Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc.
Max-Forwards: 70.
From: "TELIAX FAX" <[url=sip:303825XXXX@70.42.223.23]sip:303825XXXX@70.42.223.23[/url]>;tag=armgX7QeNQ94N.
To: <[url=sip:317376XXXX@70.88.65.1:50085]sip:317376XXXX@70.88.65.1:50085[/url]>;tag=as78a21a0c.
Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.
CSeq: 106878444 ACK.
Contact: <[url=sip:mod_sofia@70.42.223.23:5060]sip:mod_sofia@70.42.223.23:5060[/url]>.
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:
Quote:
doh,
I keep doing that sorry.

apply-nat-acl not apply_nat_acl



On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com (daldworth@teliax.com)> wrote:
Quote:
Yes. Below are settings that have been persistent through recent testing. 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:

Quote:
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 <daldworth@teliax.com (daldworth@teliax.com)> wrote:
Quote:
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 UDP header?

David


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

Quote:
0.0.0.0/0 should match all IP space.

/b

On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:

Quote:
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. What would be
the correct acl to match all IP's?

David


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--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org







_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




Back to top
anthony.minessale at g...
Guest
PostPosted: Wed Nov 19, 2008 9:13 am    Post subject: [Freeswitch-users] Wrong IP on ACK? Reply with quote
brian is out of town today
can you ping me on irc and send me to login credential and i can try to have a look.

again, you understand that we are not doing anything wrong here and what I am trying to do is a hack for your sake right?


On Wed, Nov 19, 2008 at 8:05 AM, David Aldworth <daldworth@teliax.com (daldworth@teliax.com)> wrote:
Quote:
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.

Help!


David

On Nov 6, 2008, at 10:55 AM, David Aldworth wrote:

Quote:
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:

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/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.
From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N.
To: <sip:317376XXXX@70.88.65.1:50085>.
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:317376XXXX@70.88.65.1>.
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/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.
From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N.
To: <sip:317376XXXX@70.88.65.1:50085>;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:317376XXXX@70.88.65.1>.
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/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.
From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N.
To: <sip:317376XXXX@70.88.65.1:50085>;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:317376XXXX@70.88.65.1>.
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:317376XXXX@70.88.65.1 SIP/2.0.
Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc.
Max-Forwards: 70.
From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N.
To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c.
Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.
CSeq: 106878444 ACK.
Contact: <sip:mod_sofia@70.42.223.23:5060>.
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:

Quote:
doh,
I keep doing that sorry.

apply-nat-acl not apply_nat_acl



On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com (daldworth@teliax.com)> wrote:
Quote:
Yes. Below are settings that have been persistent through recent testing. 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:

Quote:
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 <daldworth@teliax.com (daldworth@teliax.com)> wrote:
Quote:
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 UDP header?

David


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

Quote:
0.0.0.0/0 should match all IP space.

/b

On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:

Quote:
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. What would be
the correct acl to match all IP's?

David


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ibc at aliax.net
Guest
PostPosted: Wed Nov 19, 2008 9:29 am    Post subject: [Freeswitch-users] Wrong IP on ACK? Reply with quote
2008/11/19 David Aldworth <daldworth@teliax.com>:

Quote:
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.

The ACK for a 200 OK (INVITE) is an in-dialog request, so it must be
sent to the SIP URI in the top most "Route" header (if there is a
proxy in the patch that added it during the INVITE) or to the SIP URI
indicated in the "Contact" of the 200 OK.
This is the right behaviour according to the SIP specs.

In your case, the 200 OK sent by Asterisk is:

-----------------------
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/UDP
70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.
User-Agent: Asterisk PBX.
Contact: <sip:317376XXXX@70.88.65.1>.
------------------

Note that the Contact says no port, so it means default SIP port => 5060.
But note that the response comes from port 50085, why? it makes no
sense, but it's not a problem in FS but in your environment or your
network.

Could you show us the INVITE sent from FS to Asterisk (including also
the real sourde/destination address). Is that IVITE sent to port 50085
or to port 5060 of Asterisk? Is there NAT somewhere between FS and
Asterisk in this scenario?


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daldworth at teliax.com
Guest
PostPosted: Wed Nov 19, 2008 9:59 am    Post subject: [Freeswitch-users] Wrong IP on ACK? Reply with quote
I'm in there.

bigdc1

On Nov 19, 2008, at 7:12 AM, Anthony Minessale wrote:
Quote:
brian is out of town today
can you ping me on irc and send me to login credential and i can try to have a look.

again, you understand that we are not doing anything wrong here and what I am trying to do is a hack for your sake right?


On Wed, Nov 19, 2008 at 8:05 AM, David Aldworth <daldworth@teliax.com (daldworth@teliax.com)> wrote:
Quote:
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.

Help!


David

On Nov 6, 2008, at 10:55 AM, David Aldworth wrote:

Quote:
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:

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/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.
From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N.
To: <sip:317376XXXX@70.88.65.1:50085>.
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:317376XXXX@70.88.65.1>.
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/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.
From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N.
To: <sip:317376XXXX@70.88.65.1:50085>;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:317376XXXX@70.88.65.1>.
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/UDP 70.42.223.23;branch=z9hG4bKU7360cS96r7Sg;received=70.42.223.23;rport=5060.
From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N.
To: <sip:317376XXXX@70.88.65.1:50085>;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:317376XXXX@70.88.65.1>.
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:317376XXXX@70.88.65.1 SIP/2.0.
Via: SIP/2.0/UDP 70.42.223.23;rport;branch=z9hG4bKvgXZ279c41Xcc.
Max-Forwards: 70.
From: "TELIAX FAX" <sip:303825XXXX@70.42.223.23>;tag=armgX7QeNQ94N.
To: <sip:317376XXXX@70.88.65.1:50085>;tag=as78a21a0c.
Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.
CSeq: 106878444 ACK.
Contact: <sip:mod_sofia@70.42.223.23:5060>.
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:

Quote:
doh,
I keep doing that sorry.

apply-nat-acl not apply_nat_acl



On Thu, Nov 6, 2008 at 8:22 AM, David Aldworth <daldworth@teliax.com (daldworth@teliax.com)> wrote:
Quote:
Yes. Below are settings that have been persistent through recent testing. 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:

Quote:
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 <daldworth@teliax.com (daldworth@teliax.com)> wrote:
Quote:
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 UDP header?

David


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

Quote:
0.0.0.0/0 should match all IP space.

/b

On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:

Quote:
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. What would be
the correct acl to match all IP's?

David


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--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
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googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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