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<div class="moz-cite-prefix">Hi ,<br>
I've added <action application="set"
data="disable_rtp_auto_adjust=true"/> to my dialplan, but not
worked.<br>
I need to explain the architecture of our environment.<br>
We have a IVVR server. <br>
FS, IVVR server and agents are all in the same LAN. The clients
are in the internet.<br>
Here is the scenario.<br>
1) A client makes call to FS.<br>
2) FS bridges to a IVVR server for playing ivvr.<br>
3) IVVR server bridges the call back to FS queue.<br>
4) Available agent pickups the call.<br>
<br>
I know it may be NAT setting problems, and I've tried to create a
new profile(template from external) for the ivvr gateway.<br>
Now the client can hear the agent, but the agent can't hear
client.<br>
<br>
Any advice?<br>
Thanks.<br>
Jimmy<br>
<br>
<br>
於 2013/3/27 下午 02:43, Godson Gera 提到:<br>
</div>
<blockquote
cite="mid:CAPTXd0qj1EYHQXhTMj0ms3354ZRpDAAZD4BGznCNBzShKiEA6g@mail.gmail.com"
type="cite"><font>If call has been successfully established and
only audio is missing then try playing with <a
moz-do-not-send="true"
href="http://wiki.freeswitch.org/wiki/Variable_disable_rtp_auto_adjust">http://wiki.freeswitch.org/wiki/Variable_disable_rtp_auto_adjust</a></font>
<div>
<br>
</div>
<div><br>
</div>
<div> --<font> </font><font><br>
</font><font>Thanks & Regards,</font><font><br>
</font><font>Godson Gera
</font><font><br>
</font></div>
<div><font><a moz-do-not-send="true" href="http://godson.in">FreeSWITCH
Consultant</a><br>
<br>
</font></div>
<font><br>
</font>
<div class="gmail_quote"><font>On Wed, Mar 27, 2013 at 9:34 AM,
Jimmy Chang </font><span dir="ltr"><font><</font><a
moz-do-not-send="true" href="mailto:chang33.tw@gmail.com"
target="_blank"><font>chang33.tw@gmail.com</font></a><font>></font></span><font>
wrote:</font><font><br>
</font>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex"><font> </font>
<div text="#000000" bgcolor="#FFFFFF"><font> </font><font
face="Arial"><font>Hi,</font><font><br>
</font><font> </font><font><br>
</font><font> I could make a call within NAT without any
problem.</font><font><br>
</font><font> When I made a call from internet, I got no
audio.</font><font><br>
</font><font> I have read the document </font><font><br>
</font><font> </font><font><br>
</font><font> </font><a moz-do-not-send="true"
href="http://wiki.freeswitch.org/wiki/NAT_Traversal"
target="_blank"><font>http://wiki.freeswitch.org/</font><font>wiki/NAT_Traversal</font></a><font><br>
</font><font> </font></font><font><br>
</font><font> </font><font face="Arial"><font face="Arial"><font>and
set the parameters.</font><font><br>
</font><font> vars.xml</font><font><br>
</font><font> <X-PRE-PROCESS cmd="set"
data="bind_server_ip=my.</font><font>server.ip"/></font><font><br>
</font><font> <X-PRE-PROCESS cmd="set"
data="external_sip_ip=my.</font><font>server.ip"/></font><font><br>
</font><font> <X-PRE-PROCESS cmd="set"
data="external_rtp_ip=my.</font><font>server.ip"/></font><font><br>
</font><font> internal.xml</font><font><br>
</font><font> <param name="ext-rtp-ip"
value="$${external_rtp_ip}"/></font><font><br>
</font><font> external.xml</font><font><br>
</font><font> <param name="ext-sip-ip"
value="$${external_sip_ip}"/></font><font><br>
</font><font> <param name="ext-rtp-ip"
value="$${external_rtp_ip}"/></font><font><br>
</font><font> </font><font><br>
</font><font> Any advice?</font><font><br>
</font><font> Thanks.</font><span class="HOEnZb"><font
color="#888888"><font><br>
</font><font> Jimmy</font><font><br>
</font><font> </font><font><br>
</font><font> </font></font></span></font></font><font>
</font></div>
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<pre wrap="">_________________________________________________________________________
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