actually its a one-way audio issue due to NAT ! <div><br><div>alok , please explain your network flow for end to end client-server-client.</div><div><br>You may need to set nat=yes for your sip peer behind NAT. If the server is behind NAT router/firewall use externip=<public.ip.of.server> field.</div>
<div>Also provide sip traces of this call.</div><div><br></div><div>Another thing to do for your learning. Execute wireshark on both softphone systems and set "sip | rtp" as filter and see where are the RTP streams going on each end !</div>
<div><br></div><div>Take a complete capture on Asterisk server by executing the command "sip set debug on" and make a call.</div><div><br></div><div>BR</div><div>Sammy</div><div><br></div><div><br></div><div><div class="gmail_quote">
On Mon, Jul 2, 2012 at 4:39 PM, Thomas Kenyon <span dir="ltr"><<a href="mailto:digium@sanguinarius.co.uk" target="_blank">digium@sanguinarius.co.uk</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="HOEnZb"><div class="h5">alok srivastava wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
dear<br>
i have configured properly asterisk. At the one end i am using x-lite soft ph and another end twinkle. call is going properly from both end but after picking the phone not able to listen other one.<br>
when i checked the port 5060 on the asterisk server it is always showing closed while i have flushed all the rules from iptables (iptables -F)<br>
<br>
PORT STATE SERVICE VERSION<br>
5060/tcp closed sip<br>
<br>
telnet localhost 5060 (could not connect)<br>
<br>
regards<br>
alok<br>
<br>
<br>
</blockquote></div></div>
SIP is only used to setup (and stop etc.) the call. The actual audio is sent via rtp.<br>
<br>
/etc/asterisk/rtp.conf<br>
<br>
Should tell which ports asterisk is using for rtp, you will need to make sure that the remote host can connect to these ports.<br>
<br>
There are lots of articles around on how to resolve this.<div class="HOEnZb"><div class="h5"><br>
<br>
<br>
<br>
--<br>
______________________________<u></u>______________________________<u></u>_________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/<u></u>mailman/listinfo/asterisk-<u></u>users</a><br>
</div></div></blockquote></div><br></div></div>