[cisco-voip] ATA : NAT problem

[Nicolas RUIZ]

Hi,

I have a Cisco ATA 186 (code 3.1.0) with NAT, I have set :

NATIP: 62.39.70.245
NATSERVER : 0
NATTIMER: 0x00000000

When, I sniff the network where is my ATA, I don't receive any SIP frames.

I have probably a NAT problem;

THIS IS A DEBUG ON MY SIP SERVER :

IP at WAN : 62.39.70.245
IP@ SIP SERVER : 80.118.128.5

U 80.118.128.5:5060 -> 62.39.70.245:5060
  INVITE sip:0170708661 at 62.39.70.245:5060;user=phone;transport=udp
SIP/2.0..Record-Route: <sip:0170708661 at 80.118.128.5;ftag=22002
  DE0-3F0;lr>..Via: SIP/2.0/UDP
80.118.128.5;branch=z9hG4bK67ae.d6569395.0..Via: SIP/2.0/UDP
80.118.128.1:5060..From: <sip:01707
  08661 at 80.118.128.1>;tag=22002DE0-3F0..To:
<sip:0170708661 at sip.vivaction.net>..Date: Tue, 14 Sep 2004 11:45:10
GMT..Call-ID: 5E0
  BE2D0-57A11D9-8505945B-39C57B0D at 80.118.128.1..Supported:
timer,100rel..Min-SE:  1800..Cisco-Guid: 1577637232-91886041-223153877
  9-969243405..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE,
OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, N
  OTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 5..Remote-Party-ID:
<sip:0170708661 at 80.118.128.1>;party=calling;screen=yes;privacy
  =off..Timestamp: 1095162310..Contact:
<sip:0170708661 at 80.118.128.1:5060>..Expires: 180..Allow-Events:
telephone-event..Content-
  Type: application/sdp..Content-Length:
291....v=0..o=CiscoSystemsSIP-GW-UserAgent 5743 1810 IN IP4
80.118.128.1..s=SIP Call..c=
  IN IP4 80.118.128.1..t=0 0..m=audio 17738 RTP/AVP 4 101 19..c=IN IP4
80.118.128.1..a=rtpmap:4 G723/8000..a=fmtp:4 annexa=no..a=
  rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:19
CN/8000..a=ptime:30..
########################
U 80.118.128.5:5060 -> 62.39.70.245:5060
  INVITE sip:0170708661 at 62.39.70.245:5060;user=phone;transport=udp
SIP/2.0..Record-Route: <sip:0170708661 at 80.118.128.5;ftag=22002
  DE0-3F0;lr>..Via: SIP/2.0/UDP
80.118.128.5;branch=z9hG4bK67ae.d6569395.0..Via: SIP/2.0/UDP
80.118.128.1:5060..From: <sip:01707
  08661 at 80.118.128.1>;tag=22002DE0-3F0..To:
<sip:0170708661 at sip.vivaction.net>..Date: Tue, 14 Sep 2004 11:45:10
GMT..Call-ID: 5E0
  BE2D0-57A11D9-8505945B-39C57B0D at 80.118.128.1..Supported:
timer,100rel..Min-SE:  1800..Cisco-Guid: 1577637232-91886041-223153877
  9-969243405..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE,
OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, N
  OTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 5..Remote-Party-ID:
<sip:0170708661 at 80.118.128.1>;party=calling;screen=yes;privacy
  =off..Timestamp: 1095162310..Contact:
<sip:0170708661 at 80.118.128.1:5060>..Expires: 180..Allow-Events:
telephone-event..Content-
  Type: application/sdp..Content-Length:
291....v=0..o=CiscoSystemsSIP-GW-UserAgent 5743 1810 IN IP4
80.118.128.1..s=SIP Call..c=
  IN IP4 80.118.128.1..t=0 0..m=audio 17738 RTP/AVP 4 101 19..c=IN IP4
80.118.128.1..a=rtpmap:4 G723/8000..a=fmtp:4 annexa=no..a=
  rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:19
CN/8000..a=ptime:30..

But when i set NATIP: 0.0.0.0

I can do make outgoing calls but, i cannnot receive call because, the proxy
sip send to the IP@ private of the ATA.

Thanks a lot

Nicolas RUIZ
os Solutions Voix-Data !

STC
Service Technique Clients
Téléphone : 0811 02 60 61
Fax :+ 33 (0) 1 47 24 74 77
stc at vivaction.com <mailto:stc at vivaction.com>

Immeuble Plein Ouest
177 av. Georges Clemenceau
92024 Nanterre - France
Tel : 0 811 02 6000
www.vivaction.com <http://www.vivaction.com>

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