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I can only make another guess. If your system is behind a firewall,
try adding 'insecure=invite' in your sip.conf's general section.<br>
<br>
To troubleshoot such cases, do a tcpdump trace like this:<br>
<br>
1. Run tcpdump on your server before making a call. Use command
"tcpdump port 5060 -s0 -w dumpfile.pcap".<br>
2. When you notice the silence problem, hangup, and stop the trace
using CTRL+C.<br>
3. Copy the dumpfile.pcap to a computer with Wireshark installed.<br>
4. Open this file in Wireshark and follow my blog at <a
href="http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/">http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/</a><br>
5. Given that you know some basics of how VoIP works over SIP, the
wireshark graph will tell you if RTP was still flowing when it was
silent. It probably is, but to which IP address.<br>
<br>
My guess is your RTP, i.e. voice date, starts flowing towards some
wrong IP address, or stop flowing, or is blocked by the router.<br>
<br>
A good solution is to put your Asterisk server in DMZ mode.<br>
<br>
There can be many other guesses, but the above is a good start.<br>
<span class="moz-txt-tag">-- <br>
</span> <br>
Zeeshan A Zakaria <br>
<br>
PBX - visionvoip.com <br>
Blog - ilovetovoip.com <br>
<br>
On 18/10/2011 10:02, Aksel Celasun wrote:
<blockquote
cite="mid:D2D971E331A88846B8F5B5B6402E747605168744D15D@Dolly.intranett.brikken.local"
type="cite">
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<p class="MsoNormal"><span style="color:#1F497D">Thank you for
replying<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D" lang="EN-US">My
sip.conf is set to no on canreinvite<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D" lang="EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="color:#1F497D" lang="EN-US"><o:p> </o:p></span></p>
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