Hi,<div>Can you check if there is any transcoding involved with these calls, or maybe some NAT handling needs to be done by asterisk so it's not stepping out of the media-path !?</div><div><br></div><div>Regards,</div>
<div>Sammy</div><div><br><br><div class="gmail_quote">On Mon, May 21, 2012 at 5:03 PM, David Wessell <span dir="ltr"><<a href="mailto:david@ringfree.biz" target="_blank">david@ringfree.biz</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I am attempting to get an asterisk server to step out of the media<br>
path, but am running into a brick wall. Can someone assist? Here's my<br>
setup..<br>
<br>
Ultimate SIP Provider ---> LCR Trunk (Asterisk 1.6) ----> PBX (Asterisk 1.8).<br>
<br>
I am attempting to get the trunk to step out of the media stream.<br>
There is no NAT involved, all machines have a public IP.<br>
<br>
In the trunk's sip.conf I have:<br>
<br>
directmedia=yes<br>
directrtpsetup=yes<br>
<br>
And on the connection to the pbx I have canreinvite=yes<br>
<br>
On the pbx I have the trunk connection set to canreinvite=yes.<br>
<br>
In the CLI on the LCR trunk I see:<br>
<br>
-- SIP/blahblah-0000000b answered SIP/1722291028-0000000a<br>
-- Native bridging SIP/1722291028-0000000a and SIP/siproutes-0000000b<br>
<br>
Which would make me think that the lcr trunk is stepping out of the<br>
media stream. However when I pull up a tcpdump in wireshark I still<br>
see a RTP connection? Can someone point me in the right direction?<br>
<br>
Thanks<br>
David<br>
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</blockquote></div><br></div>