[Freeswitch-users] Wrong IP on ACK?
Anthony Minessale
anthony.minessale at gmail.com
Thu Nov 6 11:38:16 PST 2008
This is svn trunk? There is no reason this should not work. it happens all
the time where this setting breaks it for people going the 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 <daldworth at teliax.com>wrote:
> 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 at 70.42.223.23<sip%3A303825XXXX at 70.42.223.23>
> >;tag=armgX7QeNQ94N.
> To: <sip:317376XXXX at 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 at 70.88.65.1 <sip%3A317376XXXX at 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 at 70.42.223.23<sip%3A303825XXXX at 70.42.223.23>
> >;tag=armgX7QeNQ94N.
> To: <sip:317376XXXX at 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 at 70.88.65.1 <sip%3A317376XXXX at 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 at 70.42.223.23<sip%3A303825XXXX at 70.42.223.23>
> >;tag=armgX7QeNQ94N.
> To: <sip:317376XXXX at 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 at 70.88.65.1 <sip%3A317376XXXX at 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 at 70.88.65.1 <sip%3A317376XXXX at 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 at 70.42.223.23<sip%3A303825XXXX at 70.42.223.23>
> >;tag=armgX7QeNQ94N.
> To: <sip:317376XXXX at 70.88.65.1:50085>;tag=as78a21a0c.
> Call-ID: 9e67419c-26cd-122c-0b81-e9d53e66cb70.
> CSeq: 106878444 ACK.
> Contact: <sip:mod_sofia at 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:
>
> 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 at teliax.com>wrote:
>
>> 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:
>>
>> 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 at teliax.com>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 UDP header?
>>>
>>> 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. What would be
>>> >> the correct acl to match all IP's?
>>> >>
>>> >> David
>>> >
>>> >
>>> > _______________________________________________
>>> > Freeswitch-users mailing list
>>> > Freeswitch-users at 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 at 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 at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> _______________________________________________
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>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 at 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 at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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