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

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 
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list 
> 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,  > <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 mailto:las...@voipfreak.net>>
> > Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> > To: OpenSIPS users mailling list  > <mailto: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
> >
> >
> >
> >
>
> --
> 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

2010-05-03 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,  <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 mailto:las...@voipfreak.net>>
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list  <mailto:users@lists.opensips.org>>
> Message-ID:
>  
>   <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  <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>  <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"  <mailto:sip%3a4572727...@voxbone.com> <mailto:sip%253a4572727...@voxbone.com>>
> > >;tag=43772.
> > To: mailto: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: .
> > 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"  <mailto:sip%3a4572727...@voxbone.com> <mailto:sip%253a4572727...@v

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,  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
>users-requ...@lists.opensips.org
>
> You can reach the person managing the list at
>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 
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list 
> Message-ID:
>
> 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 
>
> >
> >
> > 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/2.0.
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
> > CSeq: 102 INVITE.
> > From: "4572727220" 
> >
> > >;tag=43772.
> > To:  <
> sip%3a1234...@11.22.33.44 >>.
> > Via: SIP/2.0/UDP 81.201.82.45:5060
> > ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
> > Max-Forwards: 69.
> > Content-Type: application/sdp.
> > Contact: .
> > 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" 
> >
> > >;tag=43772.
> > To:  <
> sip%3a1234...@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: .
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
> > CSeq: 102 INVITE.
> > From: "4572727220" 
> >
> > >;tag=43772.
> > To:  <
> sip%3a1234...@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: .
> > 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: .
> > Contact: .
> > To:  <
> sip%3a1234...@11.22.33.44 >>;tag=611cee1e

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

2010-04-29 Thread Laszlo

gt; 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" 
> >;tag=43772.
> To: >;tag=611cee1e.
> Via: SIP/2.0/UDP 81.201.82.45:5060
> ;branch=z9hG4bKb10281cafd7849f7565fbfda2b5951fd.
> Max-Forwards: 69.
> Contact: .
> Route: .
> 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" 
> >;tag=43772.
> To: >;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: .
> 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: .
> Contact: .
> To: "4572727220"
> >;tag=43772.
> From: >;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:  ;rinstance=25bfe05618433c26;nat=yes>.
> To: "4572727220"
> >;tag=43772.
> From: >;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" 
> >;tag=43772.
> From: >;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" 
> >;tag=43772.
> From: >;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 
>> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
>> To: OpenSIPS users mailling list 
>> Message-ID:
>>
>> 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" escreveu:
>>
>>
>> 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

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

2010-04-29 Thread Ahmed Munir

.
Max-Forwards: 68.
Contact: .
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: .
Contact: .
To: "4572727220"
>;tag=43772.
From: >;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: .
To: "4572727220"
>;tag=43772.
From: >;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" 
>;tag=43772.
From: >;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" 
>;tag=43772.
From: >;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 
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list 
> Message-ID:
>
> 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" escreveu:
>
>
> 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)
>
>
>
>
>
> Kindly help me out with this problem, in which other section Natting is
> required?(or am I missing something in the configuration?)  Please assist
> me
> on it.
> --
> Regards,
>
> Ahmed Munir
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> -- next part --
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> --
>
> ___
> Users mailing list
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> End of Users Digest, Vol 21, Issue 146
> **
>



-- 
Regards,

Ahmed Munir





-- 
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

2010-04-29 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" escreveu:

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=$avp(s:enum_tld)")
modparam("auth_aaa","aaa_url","radius:/usr/etc/radiusclient-ng/radiusclient.conf")
modparam("auth", "rpid_prefix", ";screen=yes;privacy=off")
modparam("auth", "rpid_avp", "$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;

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

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

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

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

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

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

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