<font color="#000066"><font><font face="tahoma,sans-serif">yes I have no control over that.</font></font></font><div><font color="#000066"><font><font face="tahoma,sans-serif"><br></font></font></font></div><div><font color="#000066"><font><font face="tahoma,sans-serif">Ok we will figure another way. Thanks </font></font></font></div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On 25 November 2012 07:10, Duncan Turnbull <span dir="ltr"><<a href="mailto:duncan@e-simple.co.nz" target="_blank">duncan@e-simple.co.nz</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto"><div class="im"><div><br><br>On 25/11/2012, at 1:23 PM, Tiago Geada <<a href="mailto:tiago.geada@gmail.com" target="_blank">tiago.geada@gmail.com</a>> wrote:<br>
<br></div><blockquote type="cite"><div><font color="#000066" face="tahoma, sans-serif">linux does sort this out and asterisk listens in both interfaces. however asterisk connects and tells remote end to send rtp back at the same IP where sip is going trough...</font><div>
<font color="#000066" face="tahoma, sans-serif"><br></font></div><div><font color="#000066" face="tahoma, sans-serif">remote end does try to send it but gets stopped in a firewall.. thus if asterisk did present a different IP to recieve RTP in its SIP header, this would not happen!</font></div>
<div class="gmail_extra"><br><br></div></div></blockquote><div><br></div></div>I think this is outside of asterisk's natural ability<div><br></div><div>You may need a proxy server in between you and the Cisco to achieve this if you can't change the firewall.</div>
<div><div><br></div><div><span style="font-family:'.HelveticaNeueUI';font-size:15px;line-height:19px;white-space:nowrap"><a href="http://forums.asterisk.org/viewtopic.php?f=1&t=84018" target="_blank">http://forums.asterisk.org/viewtopic.php?f=1&t=84018</a></span></div>
<div><br></div><div>Have you tried making the preferred route to these addresses go out eth1, thus getting the required address?</div><div><br></div><div>Ultimately seems odd the firewall allows access in but not out, guessing you have no control over that? </div>
<div><br></div><div>Good luck</div><div><br></div><div>Cheers Duncan </div><div><div class="h5"><div><br></div><br><blockquote type="cite"><div><div class="gmail_extra"><div class="gmail_quote">On 23 November 2012 19:39, Duncan Turnbull <span dir="ltr"><<a href="mailto:duncan@e-simple.co.nz" target="_blank">duncan@e-simple.co.nz</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word"><br><div><div><div><div>On 24/11/2012, at 2:19 AM, Tiago Geada <<a href="mailto:tiago.geada@gmail.com" target="_blank">tiago.geada@gmail.com</a>> wrote:</div>
<br><blockquote type="cite"><font color="#000066"><font><font face="tahoma,sans-serif">Hello Folks, I am looking for a way that makes asterisk tell remote SIP party that the IP where they will send RTP is not the same as the one I am comunicating via SIP</font></font></font><div>
<font color="#000066"><font><font face="tahoma,sans-serif"><br></font></font></font></div><div><font color="#000066"><font><font face="tahoma,sans-serif">Can this be done anyhow?</font></font></font></div><div><font color="#000066"><font><font face="tahoma,sans-serif"><br>
</font></font></font></div><div><font color="#000066"><font><font face="tahoma,sans-serif">I can try and explain:</font></font></font></div><div><font color="#000066"><font><font face="tahoma,sans-serif"><br></font></font></font></div>
<div><font color="#000066"><font><font face="tahoma,sans-serif">We have placed a asterisk box in our partners office.</font></font></font></div><div><br></div><div><font color="#000066" face="tahoma, sans-serif">It has eth0 with IP 172.16.1.10 and eth1: 10.34.18.250</font></div>
<div><font color="#000066" face="tahoma, sans-serif"><br></font></div><div><font color="#000066" face="tahoma, sans-serif">linux has its routes set so it can comunicate with several networks in their offices.</font></div>
<div><font color="#000066" face="tahoma, sans-serif"><br></font></div><div><font color="#000066" face="tahoma, sans-serif">now there is a cisco call manager that we need to communicate with. Normally via our IP 172.16.1.10, however seems that this cisco uses some sort of 'directmedya=yes' and sets both ends speaking RTP with themselves.</font></div>
<div><font color="#000066" face="tahoma, sans-serif"><br></font></div><div><font color="#000066" face="tahoma, sans-serif">There are some extensions in cisco that have a network <a href="http://10.134.0.0/16" target="_blank">10.134.0.0/16</a> that we can only comunicate via eth1</font></div>
<div><font color="#000066" face="tahoma, sans-serif"><br></font></div><div><font color="#000066" face="tahoma, sans-serif">thus when calling cisco (always via eth0) sometimes we need to say that OUR IP to recieve RTP is not 172.16.1.10, but 10.34.18.250</font></div>
<div><font color="#000066" face="tahoma, sans-serif"><br></font></div></blockquote><div><br></div></div></div><div>This is a routing issue, not asterisk I think. You are saying you route to cisco via eth0, it sets up connections to its end points and then drops out of the media flow, but the end points have no route to the eth0 address so they fail</div>
<div><br></div>Linux usually sorts this out and asterisk replies on the address of the interface it sends out with. So for the most part the response in my experience if its going out eth1 should use the eth1 ip address.</div>
<div><br></div><div>If you can get to it via eth0 and thats the preferred route then it will have the eth0 address. If so why can't you change your routing table to use eth1 when you need to go to the cisco then you will have the right address and the far extensions can respond to you correctly</div>
<div><br></div><div>Or change the cisco network endpoints so they can successfully access your address on eth0</div><div><br></div><div><br><blockquote type="cite"><div><font color="#000066" face="tahoma, sans-serif">can this be done?</font></div>
<span><font color="#888888">
--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></font></span></blockquote>
</div><br></div><br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br></div>
</div></blockquote><blockquote type="cite"><div><span>--</span><br><span>_____________________________________________________________________</span><br><span>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --</span><br>
<span>New to Asterisk? Join us for a live introductory webinar every Thurs:</span><br><span> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a></span><br><span></span><br>
<span>asterisk-users mailing list</span><br><span>To UNSUBSCRIBE or update options visit:</span><br><span> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span></div>
</blockquote></div></div></div></div><br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br></div>