Hi,<div>I can see you've tried calling route[NATMANAGE] just before the route[TOVOICEMAIL] ! and that didn't work. Can you paste your configuration as well as a SIP trace for a voicemail call ! some logs of the same calls will help too.</div>
<div><br></div><div>Regards,</div><div>Sammy</div><div><br><br><div class="gmail_quote">On Wed, May 9, 2012 at 9:10 PM, <span dir="ltr"><<a href="mailto:x-kamailio@sidell.org" target="_blank">x-kamailio@sidell.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Greetings,<br>
<br>
Here's another problem I'm having with kamailio 3.2 and the standard<br>
kamailio.cfg script.<br>
<br>
If the calling device is NATed, everything works fine if the original<br>
call gets connected. That is, the INVITE sent to the called device has<br>
the correct NAT fixups applied.<br>
<br>
But if the called device fails to answer and the script runs<br>
route[TOVOICEMAIL], the call connects, but the INVITE sent to the<br>
voicemail server doesn't have the NAT fixup applied. The result is<br>
that the audio is connected in only one direction.<br>
<br>
It would appear that some rtpproxy function needs to get called to<br>
apply the fixups prior to sending the INVITE to the voicemail server.<br>
I've tried adding calls to route(NATMANAGE) at various places, but to<br>
no avail.<br>
<br>
Any ideas?<br>
<span class="HOEnZb"><font color="#888888"><br>
--<br>
Mark Sidell<br>
Partner<br>
Forte, Inc.<br>
919-942-7068<br>
fax 919-969-2844<br>
<a href="http://www.forteinc.com" target="_blank">www.forteinc.com</a><br>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
</font></span></blockquote></div><br></div>