Open the pcap file in wireshark and see if the destination port in the SDP matches the actual port number to which the incoming RTP packets are being sent. My guess is the router/NAT device is messing with your RTP traffic.<div>
<br></div><div>-MC<br><br><div class="gmail_quote">On Mon, Sep 12, 2011 at 6:31 PM, Chad Vogel <span dir="ltr"><<a href="mailto:cvogel@lyonl.com">cvogel@lyonl.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div style="word-wrap:break-word">
I was able to confirm that i have one way and two way audio now (I can see it in wireshark), it seems like the echo application doesn't work correctly, it will not send out audio. when i use <span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px"><</span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px"><span style="color:rgb(178, 34, 33)">action</span></span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px"> </span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px"><span style="color:rgb(255, 27, 24)">application</span></span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px">=</span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px"><span style="color:rgb(0, 0, 0)">"</span></span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px">delay_echo</span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px"><span style="color:rgb(0, 0, 0)">"</span></span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px"> </span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px"><span style="color:rgb(255, 27, 24)">data</span></span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px">=</span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px"><span style="color:rgb(0, 0, 0)">"</span></span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px">1000</span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px"><span style="color:rgb(0, 0, 0)">"</span></span><span style="color:rgb(44, 38, 249);font-family:Consolas;font-size:10px">/> </span>in
the dialplan it does seems to send out rtc traffic however the audio is static noise. when i use <span style="color:rgb(255, 27, 24);font-family:Consolas;font-size:10px"><span style="color:rgb(44, 38, 249)"><</span></span><span style="color:rgb(255, 27, 24);font-family:Consolas;font-size:10px"><span style="color:rgb(178, 34, 33)">action</span></span><span style="color:rgb(255, 27, 24);font-family:Consolas;font-size:10px"><span style="color:rgb(44, 38, 249)"> </span></span><span style="color:rgb(255, 27, 24);font-family:Consolas;font-size:10px">application</span><span style="color:rgb(255, 27, 24);font-family:Consolas;font-size:10px"><span style="color:rgb(44, 38, 249)">=</span></span><span style="color:rgb(255, 27, 24);font-family:Consolas;font-size:10px"><span style="color:rgb(0, 0, 0)">"</span></span><span style="color:rgb(255, 27, 24);font-family:Consolas;font-size:10px"><span style="color:rgb(44, 38, 249)">echo</span></span><span style="color:rgb(255, 27, 24);font-family:Consolas;font-size:10px"><span style="color:rgb(0, 0, 0)">"</span></span><span style="color:rgb(255, 27, 24);font-family:Consolas;font-size:10px"><span style="color:rgb(44, 38, 249)"> /></span></span>
it doest seem to work. any thoughts on why cant I echo test and why delay_echo sends out static noise. I tried to use the record app it creates a wav file of 44 KB but doesn't grow beyond.
<div><div></div><div class="h5"><div><br>
<div>
<div> <br>
<div>
<div>On Sep 12, 2011, at 7:18 PM, Michael Collins wrote:</div>
<br>
<blockquote type="cite">Can you confirm if you have one-way audio? That is, an echo test won't tell you if you have one-way audio. A simple way to do this is to use the record app. It will play a file (whatever you choose, like "record at the tone..." - there
are tons of sound files you can try) and then record the audio from the caller. Another test to do is to get a pcap of that call so you can analyze it in wireshark. If you have RTP going in both directions to/from the FS box then that indicates a NAT issue...
<div><br>
</div>
<div>-MC<br>
<br>
<div class="gmail_quote">On Sun, Sep 11, 2011 at 7:00 PM, Chad Vogel <span dir="ltr">
<<a href="mailto:cvogel@lyonl.com" target="_blank">cvogel@lyonl.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">Level 3 uses port 5070 for their sip server but requires port 5060 to be used on our side; I made changes to the vars.xml config file to support this. I was just able to get the server to answer a call by changing the register
value to false and adding an entry in the ACL config for the Level 3 server however now it seems that the audio isn't working correctly. I'm using a simple dial plan to echo the audio back but all I get is dead air.
<div><br>
</div>
<div>
<div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;color:rgb(44, 38, 249)">
<<span style="color:#b22221">extension</span> <span style="color:#ff1b18">name</span>=<span style="color:#000000">"</span>level3<span style="color:#000000">"</span>></div>
<div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;color:rgb(44, 38, 249)">
<<span style="color:#b22221">condition</span> <span style="color:#ff1b18">field</span>=<span style="color:#000000">"</span>destination_number<span style="color:#000000">"</span>
<span style="color:#ff1b18">expression</span>=<span style="color:#000000">"</span>^13127567000$<span style="color:#000000">"</span>></div>
<div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;color:rgb(44, 38, 249)">
<<span style="color:#b22221">action</span> <span style="color:#ff1b18">application</span>=<span style="color:#000000">"</span>answer<span style="color:#000000">"</span>/></div>
<div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;color:rgb(44, 38, 249)">
<<span style="color:#b22221">action</span> <span style="color:#ff1b18">application</span>=<span style="color:#000000">"</span>echo<span style="color:#000000">"</span> /></div>
<div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;color:rgb(178, 34, 33)">
<span style="color:#2c26f9"> </</span>condition<span style="color:#2c26f9">></span></div>
<div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;color:rgb(178, 34, 33)">
<span style="color:#2c26f9"> </</span>extension<span style="color:#2c26f9">></span></div>
<div>
<div></div>
<div>
<div><span style="color:#2c26f9"><br>
</span></div>
<div><br>
</div>
<div><br>
<div>
<div>On Sep 11, 2011, at 7:47 PM, Nandy Dagondon wrote:</div>
<br>
<blockquote type="cite">in addition to peter's advise, take a look at the SIP port 5070. FS is using port 5080 for the external SIP profile. modify the port number at "external.xml" then delete the port numbers in your proxy settings.<br>
<br>
<div class="gmail_quote">On Sun, Sep 11, 2011 at 6:36 PM, Peter Olsson <span dir="ltr">
<<a href="mailto:peter.olsson@visionutveckling.se" target="_blank">peter.olsson@visionutveckling.se</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">
You're not giving us much information here. Please post exactly what doesn't work, and also pastebin the actual logs from FreeSWITCH.<br>
<br>
/Peter<br>
________________________________________<br>
Från: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">
freeswitch-users-bounces@lists.freeswitch.org</a> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] f&#246;r Chad Vogel [<a href="mailto:cvogel@lyonl.com" target="_blank">cvogel@lyonl.com</a>]<br>
Skickat: den 11 september 2011 03:28<br>
Till: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
Ämne: [Freeswitch-users] external sip profile<br>
<div>
<div></div>
<div><br>
hello,<br>
<br>
I'm trying to switch from asterisk to freeswitch; however i'm wondering how can I create a sip profile because the sip profile i created doesn't seem to function with level 3.<br>
<br>
here is the sip profile i created that isn't working:<br>
<br>
<include><br>
<gateway name="Level3"><br>
<param name="username" value="USERNAME"/><br>
<param name="password" value="PASSWORD"/><br>
<param name="proxy" value="<a href="http://4.55.35.60:5070/" target="_blank">4.55.35.60:5070</a>"/><br>
<param name="expire-seconds" value="3600"/><br>
<param name="register-transport" value="udp"/><br>
<param name="register" value="true"/><br>
</gateway><br>
</include><br>
<br>
here is my asterisk profile (it works):<br>
<br>
[level3_out]<br>
type=peer<br>
nat=no<br>
host=4.55.35.60<br>
username=***Username***<br>
secret=***Password***<br>
dtmfmode=rfc2833<br>
port=5070<br>
<br>
[level3_in]<br>
nat=no<br>
insecure=very<br>
dtmfmode=rfc2833<br>
disallow=all<br>
context=from-trunk<br>
canreinvite=no<br>
allow=ulaw&alaw<br>
host=4.55.35.60<br>
type=peer<br>
port=5070<br>
<br>
How can I create a sip profile that will function the same in freeswitch?<br>
<br>
</div>
</div>
!DSPAM:4e6c82af32761635315745!<br>
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