Have you tried turning off icmp redirect on your router?<br><br><br><div><span class="gmail_quote">On 6/6/06, <b class="gmail_sendername">Brett N</b> <<a href="mailto:brettlist@nemeroff.com">brettlist@nemeroff.com</a>> wrote:
</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi All,<br>I'm having a really weird can reinvite issue. I've been banging my head<br>
around on this for days now..<br><br><br>I have two asterisk servers. One at <a href="http://172.20.0.11">172.20.0.11</a> One at <a href="http://172.20.2.5">172.20.2.5</a><br><br><a href="http://172.20.0.11">172.20.0.11</a>
is a hosted box and serves multiple offices<br><a href="http://172.20.2.5">172.20.2.5</a> is a box on site at a customer's office.<br><br><br>A phone at <a href="http://172.20.128.10">172.20.128.10</a> makes a call using server
<a href="http://172.20.0.11">172.20.0.11</a> to a phone<br>at <a href="http://172.20.2.80">172.20.2.80</a> via server <a href="http://172.20.2.5">172.20.2.5</a>:<br><br>Phone A-->asterisk A----->SER----->asterisk B--->PhoneB
<br><br>All devices all have ip connectivity (No Firewalls! No Natting) to each<br>other. so phone a can ping phone b and server b, etc, etc, etc..<br><br><br>Can reinvite is enabled on both the ser connection (on both sides) and for
<br>both phones..<br><br>Making a call from phone A to phone B works great.. Except you can hear a<br>pop when the reinvite happens. After the call is connected Phone B can<br>transfer the phone just fine.. However if phone A (the originator) tries
<br>to transfer FIRST (either to the pstn via SER or to another local<br>extension on asterisk A) the call will have 0 way audio. If the call is<br>transfered back, there will be one way audio.<br><br>It seems this is Always how it is, over and over.. The Originator Cannot
<br>transfer the call first. I THINK if the destination transfers first, THEN<br>the originator can transfer..<br><br>I've checked netmasks, ips, gateways, etc, etc.. The SDP on the reinvites<br>looks ok..<br><br>No Nat, no funny business here.. just IP routing..
<br><br>Any ideas?<br>-Brett<br><br><br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list
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-- <br>Jim Freeze