Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-05-03 Thread Ahmed Munir 
Hi,

Thanks for replying. Can you please check my configuration of OpenSIPs what
I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146.

Please point out in which section do I required to add force_rtp_proxy(),
because I already configured Nat on it. kindly advise me soon.

On Fri, Apr 30, 2010 at 11:35 AM, users-requ...@lists.opensips.org wrote:

 Send Users mailing list submissions to
users@lists.opensips.org

 To subscribe or unsubscribe via the World Wide Web, visit
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 or, via email, send a message with subject or body 'help' to
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 You can reach the person managing the list at
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 When replying, please edit your Subject line so it is more specific
 than Re: Contents of Users digest...


 Today's Topics:

   1. Re: NAT Problem using Nat helper (Laszlo)


 --

 Message: 1
 Date: Fri, 30 Apr 2010 08:35:00 +0200
 From: Laszlo las...@voipfreak.net
 Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
 To: OpenSIPS users mailling list users@lists.opensips.org
 Message-ID:
r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Hi Ahmed,

 As you can see, the other party gets local ip in SDP

 c=IN IP4 192.168.0.168.

 You can try to play with flags:
 http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028

 -Laszlo



 2010/4/30 Ahmed Munir ahmedmunir...@gmail.com

 
 
  Hi.
 
  Thanks for your reply, the traces are metioned below;
 
  U 203.215.176.22:55134 - 11.22.33.44:5060
  .
  .
  ..
 
  U 81.201.82.45:5060 - 11.22.33.44:5060
  INVITE sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 
 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44 SIP/2.0.
  Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
  CSeq: 102 INVITE.
  From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 sip%3a4572727...@voxbone.com sip%253a4572727...@voxbone.com
  ;tag=43772.
  To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 
 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44.
  Via: SIP/2.0/UDP 81.201.82.45:5060
  ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
  Max-Forwards: 69.
  Content-Type: application/sdp.
  Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
  User-Agent: Vox Callcontrol.
  Content-Length: 210.
  .
  v=0.
  o=root 13293 13293 IN IP4 81.201.82.146.
  s=session.
  c=IN IP4 81.201.82.146.
  t=0 0.
  m=audio 11458 RTP/AVP 8 0.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:0 PCMU/8000.
  a=silenceSupp:off - - - -.
  a=ptime:20.
  a=sendrecv.
 
 
  U 11.22.33.44:5060 - 81.201.82.45:5060
  SIP/2.0 100 Giving a try.
  Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
  CSeq: 102 INVITE.
  From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 sip%3a4572727...@voxbone.com sip%253a4572727...@voxbone.com
  ;tag=43772.
  To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 
 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44.
  Via: SIP/2.0/UDP 81.201.82.45:5060
  ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0;rport=5060.
  Server: OpenSIPS (1.6.1-notls (i386/linux)).
  Content-Length: 0.
  .
 
 
  U 11.22.33.44:5060 - 203.215.176.22:55134
  INVITE sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0.
  Record-Route: sip:11.22.33.44;lr=on.
  Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
  CSeq: 102 INVITE.
  From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 sip%3a4572727...@voxbone.com sip%253a4572727...@voxbone.com
  ;tag=43772.
  To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 
 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44.
  Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
  Via: SIP/2.0/UDP 81.201.82.45:5060
 
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
  Max-Forwards: 68.
  Content-Type: application/sdp.
  Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
  User-Agent: Vox Callcontrol.
  Content-Length: 210.
  P-hint: usrloc applied.
  .
  v=0.
  o=root 13293 13293 IN IP4 81.201.82.146.
  s=session.
  c=IN IP4 81.201.82.146.
  t=0 0.
  m=audio 11458 RTP/AVP 8 0.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:0 PCMU/8000.
  a=silenceSupp:off - - - -.
  a=ptime:20.
  a=sendrecv.
 
 
  U 203.215.176.22:55134 - 11.22.33.44:5060
  SIP/2.0 180 Ringing.
  Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
  Via: SIP/2.0/UDP 81.201.82.45:5060
 
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
  Record-Route: sip:11.22.33.44;lr.
  Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26.
  To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44 
 sip%3a1234...@11.22.33.44 sip%253a1234...@11.22.33.44;tag=611cee1e.
  From: 4572727220sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 sip%3a4572727

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-05-03 Thread Bogdan-Andrei Iancu 
Hi Ahmed,

as a hint, probably you do not handle correctly the case when only the 
callee is nated (caller is public) - for such cases, to see if rtpproxy 
is needed, after the lookup(location) the nat_bflag will will 
automatically set if the callee location is nated - you can use that 
flag to detect the nated callee and to do the nat fixups - force rtpp 
and fix the 200 ok from the callee (SDP and contact).

Regards,
Bogdan

Ahmed Munir wrote:
 Hi,

 Thanks for replying. Can you please check my configuration of OpenSIPs 
 what I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146.

 Please point out in which section do I required to add 
 force_rtp_proxy(), because I already configured Nat on it. kindly 
 advise me soon.

 On Fri, Apr 30, 2010 at 11:35 AM, users-requ...@lists.opensips.org 
 mailto:users-requ...@lists.opensips.org wrote:

 Send Users mailing list submissions to
users@lists.opensips.org mailto:users@lists.opensips.org

 To subscribe or unsubscribe via the World Wide Web, visit
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 or, via email, send a message with subject or body 'help' to
users-requ...@lists.opensips.org
 mailto:users-requ...@lists.opensips.org

 You can reach the person managing the list at
users-ow...@lists.opensips.org
 mailto:users-ow...@lists.opensips.org

 When replying, please edit your Subject line so it is more specific
 than Re: Contents of Users digest...


 Today's Topics:

   1. Re: NAT Problem using Nat helper (Laszlo)


 --

 Message: 1
 Date: Fri, 30 Apr 2010 08:35:00 +0200
 From: Laszlo las...@voipfreak.net mailto:las...@voipfreak.net
 Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
 To: OpenSIPS users mailling list users@lists.opensips.org
 mailto:users@lists.opensips.org
 Message-ID:
  
  r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
 mailto:r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Hi Ahmed,

 As you can see, the other party gets local ip in SDP

 c=IN IP4 192.168.0.168.

 You can try to play with flags:
 http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028

 -Laszlo



 2010/4/30 Ahmed Munir ahmedmunir...@gmail.com
 mailto:ahmedmunir...@gmail.com

 
 
  Hi.
 
  Thanks for your reply, the traces are metioned below;
 
  U 203.215.176.22:55134 http://203.215.176.22:55134 -
 11.22.33.44:5060 http://11.22.33.44:5060
  .
  .
  ..
 
  U 81.201.82.45:5060 http://81.201.82.45:5060 -
 11.22.33.44:5060 http://11.22.33.44:5060
  INVITE sip:1234...@11.22.33.44
 mailto:sip%3a1234...@11.22.33.44 sip%3a1234...@11.22.33.44
 mailto:sip%253a1234...@11.22.33.44 SIP/2.0.
  Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45
 mailto:nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
  CSeq: 102 INVITE.
  From: 4572727220 sip:4572727...@voxbone.com
 mailto:sip%3a4572727...@voxbone.comsip%3a4572727...@voxbone.com
 mailto:sip%253a4572727...@voxbone.com
  ;tag=43772.
  To: sip:1234...@11.22.33.44 mailto:sip%3a1234...@11.22.33.44
 sip%3a1234...@11.22.33.44 mailto:sip%253a1234...@11.22.33.44.
  Via: SIP/2.0/UDP 81.201.82.45:5060 http://81.201.82.45:5060
  ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
  Max-Forwards: 69.
  Content-Type: application/sdp.
  Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
  User-Agent: Vox Callcontrol.
  Content-Length: 210.
  .
  v=0.
  o=root 13293 13293 IN IP4 81.201.82.146.
  s=session.
  c=IN IP4 81.201.82.146.
  t=0 0.
  m=audio 11458 RTP/AVP 8 0.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:0 PCMU/8000.
  a=silenceSupp:off - - - -.
  a=ptime:20.
  a=sendrecv.
 
 
  U 11.22.33.44:5060 http://11.22.33.44:5060 -
 81.201.82.45:5060 http://81.201.82.45:5060
  SIP/2.0 100 Giving a try.
  Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45
 mailto:nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
  CSeq: 102 INVITE.
  From: 4572727220 sip:4572727...@voxbone.com
 mailto:sip%3a4572727...@voxbone.comsip%3a4572727...@voxbone.com
 mailto:sip%253a4572727...@voxbone.com
  ;tag=43772.
  To: sip:1234...@11.22.33.44 mailto:sip%3a1234...@11.22.33.44
 sip%3a1234...@11.22.33.44 mailto:sip%253a1234...@11.22.33.44.
  Via: SIP/2.0/UDP 81.201.82.45:5060 http://81.201.82.45:5060
  ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0;rport=5060.
  Server: OpenSIPS (1.6.1-notls (i386/linux)).
  Content-Length: 0.
  .
 
 
  U 11.22.33.44:5060 http://11.22.33.44:5060 -
 203.215.176.22:55134 http://203.215.176.22:55134
  INVITE sip:4

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-05-03 Thread Ahmed Munir 
Hi,

Thanks for supporting me, really appreciated your help.


 Date: Mon, 03 May 2010 12:39:55 +0300
 From: Bogdan-Andrei Iancu bog...@voice-system.ro
 Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
 To: OpenSIPS users mailling list users@lists.opensips.org
 Message-ID: 4bde99eb.9090...@voice-system.ro
 Content-Type: text/plain; charset=ISO-8859-1; format=flowed

 Hi Ahmed,

 as a hint, probably you do not handle correctly the case when only the
 callee is nated (caller is public) - for such cases, to see if rtpproxy
 is needed, after the lookup(location) the nat_bflag will will
 automatically set if the callee location is nated - you can use that
 flag to detect the nated callee and to do the nat fixups - force rtpp
 and fix the 200 ok from the callee (SDP and contact).

 Regards,
 Bogdan

 Ahmed Munir wrote:
  Hi,
 
  Thanks for replying. Can you please check my configuration of OpenSIPs
  what I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146.
 
  Please point out in which section do I required to add
  force_rtp_proxy(), because I already configured Nat on it. kindly
  advise me soon.
 
  On Fri, Apr 30, 2010 at 11:35 AM, users-requ...@lists.opensips.org
  mailto:users-requ...@lists.opensips.org wrote:
 
  Send Users mailing list submissions to
 users@lists.opensips.org mailto:users@lists.opensips.org
 
  To subscribe or unsubscribe via the World Wide Web, visit
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
  or, via email, send a message with subject or body 'help' to
 users-requ...@lists.opensips.org
  mailto:users-requ...@lists.opensips.org
 
  You can reach the person managing the list at
 users-ow...@lists.opensips.org
  mailto:users-ow...@lists.opensips.org
 
  When replying, please edit your Subject line so it is more specific
  than Re: Contents of Users digest...
 
 
  Today's Topics:
 
1. Re: NAT Problem using Nat helper (Laszlo)
 
 
 
 --
 
  Message: 1
  Date: Fri, 30 Apr 2010 08:35:00 +0200
  From: Laszlo las...@voipfreak.net mailto:las...@voipfreak.net
  Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
  To: OpenSIPS users mailling list users@lists.opensips.org
  mailto:users@lists.opensips.org
  Message-ID:
 
   r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
  mailto:
 r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com
  Content-Type: text/plain; charset=iso-8859-1
 
  Hi Ahmed,
 
  As you can see, the other party gets local ip in SDP
 
  c=IN IP4 192.168.0.168.
 
  You can try to play with flags:
 
 http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028
 
  -Laszlo
 
 
 
 

 --
 Bogdan-Andrei Iancu
 www.voice-system.ro




 --

 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users


 End of Users Digest, Vol 22, Issue 13
 *




-- 
Regards,

Ahmed Munir
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-04-30 Thread Ahmed Munir 
-Length: 130.
.
v=0.
o=- 2 2 IN IP4 192.168.0.168.
s=CounterPath X-Lite 3.0.
c=IN IP4 192.168.0.168.
t=0 0.
m=audio 1876 RTP/AVP 8 0.
a=sendrecv.


U 81.201.82.45:5060 - 11.22.33.44:5060
ACK sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26;nat=yes
SIP/2.0.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 ACK.
From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
;tag=43772.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Via: SIP/2.0/UDP 81.201.82.45:5060
;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd.
Max-Forwards: 69.
Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
Route: sip:11.22.33.44;lr.
User-Agent: Vox Callcontrol.
Content-Length: 0.
.


U 11.22.33.44:5060 - 203.215.176.22:55134
ACK sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 102 ACK.
From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
;tag=43772.
To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.2.
Via: SIP/2.0/UDP 81.201.82.45:5060
;rport=5060;received=81.201.82.45;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd.
Max-Forwards: 68.
Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
User-Agent: Vox Callcontrol.
Content-Length: 0.
.


U 11.22.33.44:5060 - 203.215.176.22:55134


U 203.215.176.22:55134 - 11.22.33.44:5060
.
.
..

U 203.215.176.22:55134 - 11.22.33.44:5060
BYE sip:4572727...@81.201.82.45:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.168:55134
;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport.
Max-Forwards: 70.
Route: sip:11.22.33.44;lr.
Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26.
To: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 2 BYE.
User-Agent: X-Lite release 1104o stamp 56125.
Reason: SIP;description=User Hung Up.
Content-Length: 0.
.



U 11.22.33.44:5060 - 81.201.82.45:5060
BYE sip:4572727...@81.201.82.45:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK529a.32cc52a.0.
Via: SIP/2.0/UDP 192.168.0.168:55134
;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
Max-Forwards: 69.
Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26;nat=yes.
To: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 2 BYE.
User-Agent: X-Lite release 1104o stamp 56125.
Reason: SIP;description=User Hung Up.
Content-Length: 0.
.


U 81.201.82.45:5060 - 11.22.33.44:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK529a.32cc52a.0,SIP/2.0/UDP
192.168.0.168:55134
;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
To: 4572727220 sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 2 BYE.
Content-Length: 0.
.


U 11.22.33.44:5060 - 203.215.176.22:55134
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.168:55134
;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
To: 4572727220 sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
;tag=43772.
From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
CSeq: 2 BYE.
Content-Length: 0.
.


Date: Thu, 29 Apr 2010 19:34:16 -0300
 From: Antonio Anderson Souza anto...@voicetechnology.com.br
 Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
 To: OpenSIPS users mailling list users@lists.opensips.org
 Message-ID:
s2o285c24cc1004291534m1deec8c4zb6c4ddb003311...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Ahmed,

 Could you send an wireshark trace to the list? It will be easier to check
 what's going wrong.

 Besta regards,

 Antonio Anderson M. Souza
 Voice Technology
 http://www.antonioams.com

 Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu:


 Hi,

 I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm
 using
 is 1.6.1 and FreeRadius verison is update 2 date. When I register 2
 sofphone, they got authenticated and authorized by radius and got
 registered sucessfully. Even I made calls between two softphone
 sucessfully(Can hear one another). The UAS configured on different network
 means hosted with public IP and my softphones are registered other and
 NATed
 network. I mapped a DID on UAS and mapped it on my one of my softphone. The
 problem I'm facing is when call coming from DID and ring my phone the
 caller
 can hear me but I can't hear the caller(one way calling issue). But not
 facing the problem

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-04-30 Thread Laszlo 
.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 From: 4572727220sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 INVITE.
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
 SUBSCRIBE, INFO.
 Content-Type: application/sdp.
 User-Agent: X-Lite release 1104o stamp 56125.
 Content-Length: 130.
 .
 v=0.
 o=- 2 2 IN IP4 192.168.0.168.
 s=CounterPath X-Lite 3.0.
 c=IN IP4 192.168.0.168.
 t=0 0.
 m=audio 1876 RTP/AVP 8 0.
 a=sendrecv.


 U 81.201.82.45:5060 - 11.22.33.44:5060
 ACK sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26;nat=yes
 SIP/2.0.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 ACK.
 From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd.
 Max-Forwards: 69.
 Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
 Route: sip:11.22.33.44;lr.
 User-Agent: Vox Callcontrol.
 Content-Length: 0.
 .


 U 11.22.33.44:5060 - 203.215.176.22:55134
 ACK sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 102 ACK.
 From: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 To: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.2.
 Via: SIP/2.0/UDP 81.201.82.45:5060
 ;rport=5060;received=81.201.82.45;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd.
 Max-Forwards: 68.
 Contact: sip:4572727...@81.201.82.45:5060;transport=udp.
 User-Agent: Vox Callcontrol.
 Content-Length: 0.
 .


 U 11.22.33.44:5060 - 203.215.176.22:55134
 

 U 203.215.176.22:55134 - 11.22.33.44:5060
 .
 .
 ..

 U 203.215.176.22:55134 - 11.22.33.44:5060
 BYE sip:4572727...@81.201.82.45:5060;transport=udp SIP/2.0.
 Via: SIP/2.0/UDP 192.168.0.168:55134
 ;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport.
 Max-Forwards: 70.
 Route: sip:11.22.33.44;lr.
 Contact: sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26.
 To: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
 ;tag=43772.
 From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 2 BYE.
 User-Agent: X-Lite release 1104o stamp 56125.
 Reason: SIP;description=User Hung Up.
 Content-Length: 0.
 .



 U 11.22.33.44:5060 - 81.201.82.45:5060
 BYE sip:4572727...@81.201.82.45:5060;transport=udp SIP/2.0.
 Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK529a.32cc52a.0.
 Via: SIP/2.0/UDP 192.168.0.168:55134
 ;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
 Max-Forwards: 69.
 Contact: sip:4...@203.215.176.22:55134
 ;rinstance=25bfe05618433c26;nat=yes.
 To: 4572727220sip:4572727...@voxbone.com sip%3a4572727...@voxbone.com
 ;tag=43772.
 From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 2 BYE.
 User-Agent: X-Lite release 1104o stamp 56125.
 Reason: SIP;description=User Hung Up.
 Content-Length: 0.
 .


 U 81.201.82.45:5060 - 11.22.33.44:5060
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK529a.32cc52a.0,SIP/2.0/UDP
 192.168.0.168:55134
 ;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
 To: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 2 BYE.
 Content-Length: 0.
 .


 U 11.22.33.44:5060 - 203.215.176.22:55134
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 192.168.0.168:55134
 ;received=203.215.176.22;branch=z9hG4bK-d8754z-cf10f40220676576-1---d8754z-;rport=55134.
 To: 4572727220 sip:4572727...@voxbone.comsip%3a4572727...@voxbone.com
 ;tag=43772.
 From: sip:1234...@11.22.33.44 sip%3a1234...@11.22.33.44;tag=611cee1e.
 Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
 CSeq: 2 BYE.
 Content-Length: 0.
 .


 Date: Thu, 29 Apr 2010 19:34:16 -0300
 From: Antonio Anderson Souza anto...@voicetechnology.com.br
 Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
 To: OpenSIPS users mailling list users@lists.opensips.org
 Message-ID:
s2o285c24cc1004291534m1deec8c4zb6c4ddb003311...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1


 Ahmed,

 Could you send an wireshark trace to the list? It will be easier to check
 what's going wrong.

 Besta regards,

 Antonio Anderson M. Souza
 Voice Technology
 http://www.antonioams.com

 Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu:


 Hi,

 I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm
 using
 is 1.6.1 and FreeRadius verison is update 2 date. When I register 2
 sofphone, they got authenticated

[OpenSIPS-Users] NAT Problem using Nat helper

2010-04-29 Thread Ahmed Munir 
Hi,

I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm using
is 1.6.1 and FreeRadius verison is update 2 date. When I register 2
sofphone, they got authenticated and authorized by radius and got
registered sucessfully. Even I made calls between two softphone
sucessfully(Can hear one another). The UAS configured on different network
means hosted with public IP and my softphones are registered other and NATed
network. I mapped a DID on UAS and mapped it on my one of my softphone. The
problem I'm facing is when call coming from DID and ring my phone the caller
can hear me but I can't hear the caller(one way calling issue). But not
facing the problem on when calling between to sip clients and also calling
from OpenSIPs to Asterisk. The configuration and cases for OpenSIPs is
listed down below;


UAC-- UAS(OpenSIPs) -
UACtwo way voice is establised
 UAC-- UAS(OpenSIPs) - Asterisk
 UACtwo way voice is establised
PSTN-- UAS(OpenSIPs)
- UAC  one way
voice is establised
(hears the dest voice)(can't hear caller
voice)


#loadmodule auth_diameter.so
loadmodule aaa_radius.so
loadmodule auth_aaa.so
loadmodule permissions.so
loadmodule nathelper.so
#Settings For
Radius-
#modparam(auth_diameter, diameter_client_host, localhost)
modparam(aaa_radius,
radius_config,/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_url,
radius:/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_flag, 2)
modparam(acc, aaa_missed_flag, 3)
modparam(acc, aaa_extra, User-Name=$Au; \
Calling-Station-Id=$from; \
Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ruri; \
Sip-RPid=$avp(s:rpid); \
Source-IP=$si; \
Source-Port=$sp; \
Canonical-URI=$avp(s:can_uri); \
Billing-Party=$avp(s:billing_party); \
Divert-Reason=$avp(s:divert_reason); \
X-RTP-Stat=$hdr(X-RTP-Stat); \
Contact=$hdr(contact); \
Event=$hdr(event); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
ENUM-TLD=$avp(s:enum_tld))
modparam(auth_aaa,aaa_url,radius:/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(auth, rpid_prefix, sip:)
modparam(auth, rpid_suffix, @11.22.33.44;screen=yes;privacy=off)
modparam(auth, rpid_avp, $avp(s:rpid))
#modparam(uri,service_type,10)
# - setting module-specific parameters ---
modparam(dispatcher, db_url, mysql://opensips:opensip...@localhost
/opensips)
modparam(permissions, db_url, mysql://opensips:opensip...@localhost
/opensips)
#- setting NAT module parameters -
modparam(nathelper,ping_nated_only,1)
modparam(nathelper, natping_interval, 30)
modparam(nathelper,natping_processes,1)
#modparam(nathelper,rtpproxy_sock,udp:127.0.0.1:7890)
modparam(nathelper,rtpproxy_sock, )
modparam(nathelper,received_avp,$avp(i:42))
#modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 6)
### Routing Logic 
# main request routing logic
route{
 if (!mf_process_maxfwd_header(10)) {
  sl_send_reply(483,Too Many Hops);
  exit;
 }

 #NAT detection
 log(# Go to Route 3 for NAT
Detection #);
 route(3);
 if (has_totag()) {
  # sequential request withing a dialog should
  # take the path determined by record-routing
  if (loose_route()) {
   if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
   } else if (is_method(INVITE)) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
   }
   # route it out to whatever destination was set by loose_route()
   # in $du (destination URI).
   route(1);
  } else {
   if ( is_method(ACK) ) {
if ( t_check_trans() ) {
 # non loose-route, but stateful ACK; must be an ACK after
 # a 487 or e.g. 404 from upstream server
 t_relay();
 exit;
} else {
 # ACK without matching transaction -
 # ignore and discard
 exit;
}
   }
   sl_send_reply(404,Not here);
  }
  exit;
 }
 #initial requests
 # CANCEL processing
 if (is_method(CANCEL))
 {
  if (t_check_trans())
   t_relay();
  exit;
 }
 t_check_trans();
 if (loose_route()) {
  xlog(L_ERR,
  Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]);
  if (!is_method(ACK))
   sl_send_reply(403,Preload Route denied);
  exit;
 }
 # record routing
 if (!is_method(REGISTER|MESSAGE))
  record_route();

 #$avp(i:27)=check_source_address(0);
 #xlog(Check Source Address from Address TABLE : $(avp(i:27))\n);
 $avp(s:checksrc) = check_source_address(0);
 
log(###\n);
 xlog(Check

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-04-29 Thread Antonio Anderson Souza 
Ahmed,

Could you send an wireshark trace to the list? It will be easier to check
what's going wrong.

Besta regards,

Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com

Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu:

Hi,

I've configured OpenSIPs with FreeRadius, the version for OpenSIPs I'm using
is 1.6.1 and FreeRadius verison is update 2 date. When I register 2
sofphone, they got authenticated and authorized by radius and got
registered sucessfully. Even I made calls between two softphone
sucessfully(Can hear one another). The UAS configured on different network
means hosted with public IP and my softphones are registered other and NATed
network. I mapped a DID on UAS and mapped it on my one of my softphone. The
problem I'm facing is when call coming from DID and ring my phone the caller
can hear me but I can't hear the caller(one way calling issue). But not
facing the problem on when calling between to sip clients and also calling
from OpenSIPs to Asterisk. The configuration and cases for OpenSIPs is
listed down below;


UAC-- UAS(OpenSIPs) -
UACtwo way voice is establised
 UAC-- UAS(OpenSIPs) - Asterisk
 UACtwo way voice is establised
PSTN-- UAS(OpenSIPs)
- UAC  one way
voice is establised
(hears the dest voice)(can't hear caller
voice)


#loadmodule auth_diameter.so
loadmodule aaa_radius.so
loadmodule auth_aaa.so
loadmodule permissions.so
loadmodule nathelper.so
#Settings For
Radius-
#modparam(auth_diameter, diameter_client_host, localhost)
modparam(aaa_radius,
radius_config,/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_url,
radius:/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_flag, 2)
modparam(acc, aaa_missed_flag, 3)
modparam(acc, aaa_extra, User-Name=$Au; \
Calling-Station-Id=$from; \
Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ruri; \
Sip-RPid=$avp(s:rpid); \
Source-IP=$si; \
Source-Port=$sp; \
Canonical-URI=$avp(s:can_uri); \
Billing-Party=$avp(s:billing_party); \
Divert-Reason=$avp(s:divert_reason); \
X-RTP-Stat=$hdr(X-RTP-Stat); \
Contact=$hdr(contact); \
Event=$hdr(event); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
ENUM-TLD=$avp(s:enum_tld))
modparam(auth_aaa,aaa_url,radius:/usr/etc/radiusclient-ng/radiusclient.conf)
modparam(auth, rpid_prefix, sip:)
modparam(auth, rpid_suffix, @11.22.33.44;screen=yes;privacy=off)
modparam(auth, rpid_avp, $avp(s:rpid))
#modparam(uri,service_type,10)
# - setting module-specific parameters ---
modparam(dispatcher, db_url, mysql://opensips:opensip...@localhost
/opensips)
modparam(permissions, db_url, mysql://opensips:opensip...@localhost
/opensips)
#- setting NAT module parameters -
modparam(nathelper,ping_nated_only,1)
modparam(nathelper, natping_interval, 30)
modparam(nathelper,natping_processes,1)
#modparam(nathelper,rtpproxy_sock,udp:127.0.0.1:7890)
modparam(nathelper,rtpproxy_sock, )
modparam(nathelper,received_avp,$avp(i:42))
#modparam(nathelper, sipping_bflag, 7)
modparam(usrloc, nat_bflag, 6)
### Routing Logic 
# main request routing logic
route{
 if (!mf_process_maxfwd_header(10)) {
  sl_send_reply(483,Too Many Hops);
  exit;
 }

 #NAT detection
 log(# Go to Route 3 for NAT
Detection #);
 route(3);
 if (has_totag()) {
  # sequential request withing a dialog should
  # take the path determined by record-routing
  if (loose_route()) {
   if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
   } else if (is_method(INVITE)) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
   }
   # route it out to whatever destination was set by loose_route()
   # in $du (destination URI).
   route(1);
  } else {
   if ( is_method(ACK) ) {
if ( t_check_trans() ) {
 # non loose-route, but stateful ACK; must be an ACK after
 # a 487 or e.g. 404 from upstream server
 t_relay();
 exit;
} else {
 # ACK without matching transaction -
 # ignore and discard
 exit;
}
   }
   sl_send_reply(404,Not here);
  }
  exit;
 }
 #initial requests
 # CANCEL processing
 if (is_method(CANCEL))
 {
  if (t_check_trans())
   t_relay();
  exit;
 }
 t_check_trans();
 if (loose_route()) {
  xlog(L_ERR,
  Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]);
  if (!is_method(ACK))
   sl_send_reply(403,Preload Route denied);
  exit;
 }
 # record routing
 if (!is_method(REGISTER|MESSAGE))
  record_route();