<div dir="ltr">Hi bluerain,<div><br></div><div>Our production setup is using OpenSIPS as proxy to many Asterisk instances via load_balancer module.</div><div><br></div><div>I can see from the SIP logs on our Asterisk servers the SIP headers sent from OpenSIPS to Asterisk still contains the original IP (From, Contact, and the bottom one Via)</div>
<div><br></div><div>Have you tried to read any of those SIP header in your Asterisk script?</div><div><br></div><div>You should also check your opensips script in /etc/opensips/opensips.cfg to see if you have rewrite SIP headers which contain the original ip.</div>
<div><br></div><div>Cheers,</div><div>Bruce</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Oct 11, 2013 at 11:38 AM, Mike Tesliuk <span dir="ltr"><<a href="mailto:mike@ultra.net.br" target="_blank">mike@ultra.net.br</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>the sugestion from Stefano is to you transport the ip information from opensips to asterisk, when you are in asterisk you get that variable and validate the customer, if just opensips will talk with asterisk so you dont need the ip address on asterisk, just in opensips, you made all validation on opensips, and send that with some kind of identification of the customer to asterisk, define as callerid or accountcode and make the next step on asterisk.<br>
<br></div>I understand what you want, maybe you should try to use some kind of redirect, you send the call to opensips, opensips check which server should receive that and so you reply with a redirect message and the call will be stablished directly with the asterisk, opensips will just show the path <br>
</div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">2013/10/10 bluerain <span dir="ltr"><<a href="mailto:frank21118@yahoo.com" target="_blank">frank21118@yahoo.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Just FYI, I tried, I insert your line in the method invite and right before<br>
the routing, Asterisk didn't seem to care. It still care about the prior<br>
Hop IP.<br>
<br>
So what I mean is that<br>
<br>
from 199.33.33.33 --> opensip 22.55.33.33 (and then I put your line) --><br>
Asterisk server.<br>
<br>
Asterisk server identified the call came from 22.55.33.33 and not<br>
199.33.33.33<br>
<br>
Frank<br>
<br>
<br>
<br>
--<br>
View this message in context: <a href="http://opensips-open-sip-server.1449251.n2.nabble.com/Opensip-as-transaprent-inbound-proxy-for-asterisk-tp7588047p7588062.html" target="_blank">http://opensips-open-sip-server.1449251.n2.nabble.com/Opensip-as-transaprent-inbound-proxy-for-asterisk-tp7588047p7588062.html</a><br>
<div><div>Sent from the OpenSIPS - Users mailing list archive at Nabble.com.<br>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>simple is good<br><a href="http://brucewang.net" target="_blank">http://brucewang.net</a><br><a href="http://twitter.com/number5" target="_blank">http://twitter.com/number5</a><br>
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