<div dir="ltr">Its going to be because you probably need to pre_answer, play a little media before you hair pin the call back out the same provider. :) playback silence_stream://1000 after pre_answer would probably fix it prior to bridge.</div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Mar 9, 2016 at 8:32 AM, Peter Steinbach <span dir="ltr"><<a href="mailto:lists@telefaks.de" target="_blank">lists@telefaks.de</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
Hello,<br>
<br>
I have the following scenario<br>
<ul>
<li>Freeswitch is behind NAT with external profile with ext_sip_ip
and ext_rtp_ip</li>
<li>Echo from an external SIP provider test inbound works with
audio</li>
<li>Calling outbound to the same external SIP provider works with
audio</li>
<li><b>Receiving a call from the external SIP provider and
bridging it to another number at the same external SIP
provider does not have audio</b></li>
<li>Receiving a call from the external SIP provider and bridging
it to a local Patton PRI works with audio</li>
</ul>
<p>I also set up another (modified NAT) profile<br>
</p>
<ul>
<li>like external profile - but with a different ext_rtp_ip </li>
<li>calling outbound to via NAT profile to another SIP provider
works with audio</li>
<li><b>Receiving a call from the external SIP provider (external
profile) and bridging it to another number </b><b><b></b>at
the another external SIP provider </b><b><b>(nat</b><b><b>
profile)</b></b> does not have audio</b></li>
<li>Tcpdump btw. does not show an attempt of Freeswitch to issue any
outbound RTP packet. I can see only SIP messages towards the
outbound SIP provider<b><br>
</b></li>
</ul>
<p>So I think, there is a problem with briding a natted external
call to another natted external call, both with ext-rtp-ip. It
tried this on an older Freeswitch (1 year old) and with a more
recent Freeswitch (1 month old), both behave the same. Same
scenario with a non-natted Freeswitch works fine. <br>
Btw. Freeswitch logs do show <br>
RTCPpacket not written<br>
right after bridginbg the call.<br>
</p>
<p>Question: Does anybody have a clue, what could cause the problem?
Maybe there is any parameter in the profiles, that may help?<span class="HOEnZb"><font color="#888888"><br>
<br>
</font></span></p><span class="HOEnZb"><font color="#888888">
<p><br>
</p>
<pre cols="72">--
With kind regards
Peter Steinbach
Telefaks Services GmbH
<a href="mailto:lists" target="_blank">mailto:lists</a> (att) <a href="http://telefaks.de" target="_blank">telefaks.de</a>
Internet: <a href="http://www.telefaks.de" target="_blank">www.telefaks.de</a>
</pre>
</font></span></div>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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