<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><meta http-equiv=Content-Type content="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 14 (filtered medium)"><style><!--
/* Font Definitions */
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:11.0pt;
font-family:"Calibri","sans-serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
span.EmailStyle17
{mso-style-type:personal-compose;
font-family:"Calibri","sans-serif";
color:windowtext;}
.MsoChpDefault
{mso-style-type:export-only;
font-size:10.0pt;}
@page WordSection1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>Hello, <o:p></o:p></p><p class=MsoNormal>I currently have a media server behind a nat firewall with calls delivered via a PSTN Trunk. I want to add a 2nd media server and route calls to either depending upon the dialed number. I’ve been trying to do this using drouting in opensips 1.10.0, but cannot get a configuration that works. <o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I started by generating the ‘trunking script’ using make menuconfig, and populated mysql to accept my PSTN trunk and route to my media server. When an incoming call arrives, it is directed to opensips, and forwarded to media server with a record-route header containing my private ip. This confuses my PSTN partner and we are unable to establish the rtp stream.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>After reviewing the mailing lists I tried setting alias and advertised_address=my public ip. Now when an incoming call arrives it is directed to opensips and forwarded to the media server with a record-route header containing my public ip. Call setup completes successfully. Call teardown initiated from PSTN trunk completes successfully. Call teardown initiated from media server fails because the media servers sends BYE to the public IP, and the NAT router does not know what to do with it (destination unreachable).<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>It seems as though the invite to my media server needs to have a record-route header with my private ip, while the ok response back to my PSTN provider needs to have a record-route header with my public ip. Is this the right approach? I’ve briefly toyed with rtpproxy and also b2bua without much luck, and was hoping this simpler solution could be made to work.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Thanks,<o:p></o:p></p><p class=MsoNormal>Tony<o:p></o:p></p></div></body></html>