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<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes"><FONT
face=Arial>Vivienne,</FONT></SPAN></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes"><FONT face=Arial>From your
config:</FONT> </SPAN></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes"> </SPAN>if
(nat_uac_test("3")){<o:p></o:p></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes">
</SPAN>if (method == "REGISTER" || !
search("^Record-Route:")){<o:p></o:p></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes">
</SPAN>log("Log: Someone trying to register from private
IP,rew<o:p></o:p></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT
face="Times New Roman">riting\n");<o:p></o:p></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-tab-count: 1">
</SPAN>fix_nated_contact(); #Rewrite contact with source
IP<o:p></o:p></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes"> </SPAN><SPAN
style="mso-spacerun: yes"> </SPAN>if
(method == "INVITE"){<o:p></o:p></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes">
</SPAN>fix_nated_sdp("1"); #Add direction=active to
SDP<o:p></o:p></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes">
</SPAN>};<o:p></o:p></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes">
</SPAN>force_rport(); # Add rport parameter to topmost
Via<o:p></o:p></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes">
</SPAN>setflag(6); # Mark as Nated<o:p></o:p></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman"><SPAN
style="mso-spacerun: yes">
</SPAN>};</FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT
face="Times New Roman"></FONT></SPAN> </P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman">Here you don't
proxy the call, you just add the direction=active. If you want to rtp
proxy the calls, you need to have a force_rtp_proxy() for the initial
INVITE. Read section 1 of the Getting Started document at ONsip.org to
understand what is supposed to happen and how to read the SDP payload and see
the rtpproxy example found in the document. </FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT face="Times New Roman">Your config will
process the OK and proxy the rtp stream from the callee, but not the
caller's.</FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-IE
style="mso-ansi-language: EN-IE"><FONT
face="Times New Roman">g-)</FONT></SPAN></P></DIV>
<DIV> </DIV>
<DIV>---- Original Message ----<BR>From: Vivienne Curran<BR>To:
serusers@lists.iptel.org<BR>Sent: Thursday, March 31, 2005 05:55 PM<BR>Subject:
[Serusers] Nathelper/Rtpproxy not working for two natted<BR>clients <BR><BR>>
Hi,<BR>> <BR>> I am having problems troubleshooting a problem I am
experiencing with<BR>> my SER configuration. I have ser 0.8.14 running with
rtpproxy and<BR>> nathelper enabled. I have two phones on the same subnet
behind nat<BR>> and I would like to make a call between the two. I want to
invoke<BR>> rtpproxy for this as they both have private address [I know
this<BR>> isn’t the most efficient way as they’re both on the same subnet but
I<BR>> can worry about that later]. <BR>>
<BR>> When I ring from the phone 1 (2092) to phone 2 (2093), 2092 can
hear<BR>> voice but 2093 can’t. When 2093 ring 2092, there’s no audio.
These<BR>> phones are Grandstream BT100’s. They have been configured to
listen<BR>> on different SIP and RTP ports. <BR>> <BR>>
2092: SIP Port: 5060<BR>> 2092: RTP Port: 5004<BR>> 2093: SIP Port:
5061<BR>> 2093: RTP Port: 5005<BR>> <BR>> I have included my ser.cfg
file the messages received on my SER<BR>> server (using ngrep SIP-q)in an
attachment. I can confirm that my<BR>> rtpproxy is working (originally I
thought it wasn’t) by using “strace<BR>> –d <rtpproxy pid> -f
–F”. I can see a signal being returned. <BR>> <BR>> Any help
would be appreciated or advise as to how I can proceed<BR>> troubleshooting.
<BR>> Kindest Regards,<BR>> Vivienne.<BR>> Send instant messages to
your online friends<BR>> http://uk.messenger.yahoo.com <BR>> <BR>>
<BR>> <BR>> _______________________________________________<BR>>
Serusers mailing list<BR>> serusers@lists.iptel.org<BR>>
http://lists.iptel.org/mailman/listinfo/serusers</DIV></BODY></HTML>