[Freeswitch-users] NAT issue with SBC

ThanhTruong ngoc217thanh at yahoo.com
Fri Sep 13 08:20:58 MSD 2013


Hi Michael Jerris,

I have tried many ways to see how asterisk reject ACK but cant see any issue in asterisk server.

this is log in asterisk server when it getting 200OK.

Hope you can help me more in this case.

<--- Reliably Transmitting (NAT) to 123.123.123.123:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bKQ6S88v04UNUgp;received=123.123.123.123;rport=5060
From: "1000" <sip:1000 at 123.123.123.123>;tag=yDvgQBjFN9teg
To: <sip:1001 at 59.120.208.196:5060>;tag=as7d0ee969
Call-ID: NDI5MzhjYjc2N2Y2NjM2OWZlYjM3YzlhODczZWRiMzQ.
CSeq: 3 INVITE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1001 at 123.123.123.124:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 304

v=0
o=root 1869340460 1869340460 IN IP4 123.123.123.124
s=Asterisk PBX 11.5.0
c=IN IP4 123.123.123.124
t=0 0
m=audio 14742 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=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34

<------------>
       > 0x1fdb630 -- Probation passed - setting RTP source address to 123.123.123.125:34156
       > 0x1fdb630 -- Probation passed - setting RTP source address to 123.123.123.125:34156

<--- SIP read from UDP:123.123.123.123:5060 --->
ACK sip:1001 at 123.123.123.124:5060 SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123;rport;branch=z9hG4bKrFK1arH8ryH3H
Max-Forwards: 70
From: "1000" <sip:1000 at 123.123.123.123>;tag=yDvgQBjFN9teg
To: <sip:1001 at 123.123.123.124>;tag=as7d0ee969
Call-ID: NDI5MzhjYjc2N2Y2NjM2OWZlYjM3YzlhODczZWRiMzQ.
CSeq: 3 ACK
Contact: <sip:1000 at 123.123.123.123:5060>
Authorization: Digest username="1000", realm="asterisk", uri="sip:1001 at 123.123.123.123", nonce="2d8e3c5a", response="afda149dd7fa2ba457ccc494c272a1a1", algorithm=MD5
Content-Length: 0


i make call from 123.123.123.125. FS server is 123.123.123.123. asterisk server is 123.123.123.124

Thank you in advance.
 
ThanhTruong


________________________________
 From: Michael Jerris <mike at jerris.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Tuesday, September 10, 2013 1:13 AM
Subject: Re: [Freeswitch-users] NAT issue with SBC
 


Then the asterisk box is not accepting the ack for some reason.


On Sep 9, 2013, at 1:57 PM, ThanhTruong <ngoc217thanh at yahoo.com> wrote:

Hi Michael Jerris,
>
>
>In Both site, ACK coming into asterisk server, then, after some second, asterisk try to send 200OK to FS and FS return ACK too.
>
>
>That is my case. 
>
>
>sorry about that if i disturb you.
>
>
>Thanks,
> 
>ThanhTruong
>
>
>
>________________________________
> From: Michael Jerris <mike at jerris.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Tuesday, September 10, 2013 12:49 AM
>Subject: Re: [Freeswitch-users] NAT issue with SBC
> 
>
>
>have you lookedat a Sip trace from both sides?  Is the ACK getting to asterisk?  Is it not accepting it for some reason?
>
>
>On Sep 9, 2013, at 1:45 PM, ThanhTruong <ngoc217thanh at yahoo.com> wrote:
>
>hi Michael Jerris,
>>
>>
>>I have sent in another email. Do you see it ?
>>
>>
>>Thanks
>>
>>
>> 
>>ThanhTruong
>>
>>
>>
>>________________________________
>> From: Michael Jerris <mike at jerris.com>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>>Sent: Tuesday, September 10, 2013 12:21 AM
>>Subject: Re: [Freeswitch-users] NAT issue with SBC
>> 
>>
>>
>>So did you do what I suggested then or not?
>>
>>
>>On Sep 9, 2013, at 1:13 PM, ThanhTruong <ngoc217thanh at yahoo.com> wrote:
>>
>>hi
>>>this is sip trace in FS server.
>>>
>>>
>>>
>>>________________________________
>>> From: Michael Jerris <mike at jerris.com>
>>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>>>Sent: Monday, September 9, 2013 7:14 PM
>>>Subject: Re: [Freeswitch-users] NAT issue with SBC
>>> 
>>>
>>>
>>>have you looked at a Sip trace from both sides?  Is the ACK getting to asterisk?  Is it not accepting it for some reason?
>>>
>>>
>>
>>_________________________________________________________________________
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>>
>>
>>
>>
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>>
>>_________________________________________________________________________
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>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
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>>
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>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>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
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>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
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




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