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<div>Thank you for replying also,</div>
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<div><font size="2" face="tahoma">I will as you and Zeeshan suggest, look at the firewall issue first, i have been suspecting</font></div>
<div><font size="2" face="tahoma">network issue, because i cannot see anything in the log, so again thanks!</font></div>
<div><font size="2" face="tahoma"></font> </div>
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<div><font size="2" face="tahoma">Best regards </font></div>
<div><font size="2" face="tahoma"></font> </div>
<div><font size="2" face="tahoma"></font> </div>
<div><font size="2" face="tahoma">Aksel</font></div>
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<font color="#000000" size="2" face="Tahoma"><b>Fra:</b> asterisk-users-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] på vegne av Sammy Govind [govoiper@gmail.com]<br>
<b>Sendt:</b> 19. oktober 2011 08:48<br>
<b>Til:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Emne:</b> Re: [asterisk-users] Problems during calls<br>
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<div>Hi,</div>
<div><br>
</div>
Call getting silenced in the middle definitely point to RTP but I think the redialling part should be considered as well. I think that Phones are loosing registrations or like Zeeshan mentioned could be getting blocked by firewall - Asterisk server's firewall
as well as any other firewall in front of server should be inspected for sessions/connections limit etc.
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<div>--</div>
<div>Regards,</div>
<div>Sammy<br>
<br>
<div class="gmail_quote">On Wed, Oct 19, 2011 at 12:27 AM, Aksel Celasun <span dir="ltr">
<<a href="mailto:aksel@abacus-it.no">aksel@abacus-it.no</a>></span> wrote:<br>
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<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US">Thank you for the reply.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US"><u></u><u></u></span> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US"><u></u><u></u></span> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US">The Asterisk is behind a firewall, but not in a dmz, been thinking of placing it in a dmz soon, maybe that will solve the problem.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US">Or else, I will try your guide with wireshark.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US"><u></u><u></u></span> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US">Thank you very much.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US"><u></u><u></u></span> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US"><u></u><u></u></span> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US">Best regards<u></u><u></u></span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US"><u></u><u></u></span> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US">Aksel<u></u><u></u></span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US"><u></u><u></u></span> </p>
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<p class="MsoNormal"><b><span style="COLOR: windowtext; FONT-SIZE: 10pt">Fra:</span></b><span style="COLOR: windowtext; FONT-SIZE: 10pt">
<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>]
<b>På vegne av</b> VisionVoIP<br>
<b>Sendt:</b> 18. oktober 2011 16:31</span></p>
<div class="im"><br>
<b>Til:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Emne:</b> Re: [asterisk-users] Problems during calls<u></u><u></u></div>
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<p class="MsoNormal"><u></u><u></u> </p>
<p class="MsoNormal">I can only make another guess. If your system is behind a firewall, try adding 'insecure=invite' in your sip.conf's general section.</p>
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<div class="h5"><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/" target="_blank">
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>-- </span><br>
<br>
Zeeshan A Zakaria <br>
<br>
PBX - <a href="http://visionvoip.com" target="_blank">visionvoip.com</a> <br>
Blog - <a href="http://ilovetovoip.com" target="_blank">ilovetovoip.com</a> <br>
<br>
On 18/10/2011 10:02, Aksel Celasun wrote: <u></u><u></u></div>
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<p class="MsoNormal"><span style="COLOR: #1f497d">Thank you for replying</span><u></u><u></u></p>
<p class="MsoNormal"><span style="COLOR: #1f497d"></span><u></u><u></u> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d"></span><u></u><u></u> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US">My sip.conf is set to no on canreinvite</span><u></u><u></u></p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US"></span><u></u><u></u> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US"></span><u></u><u></u> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d" lang="EN-US"></span><u></u><u></u> </p>
<p class="MsoNormal"><span style="FONT-FAMILY: 'Times New Roman','serif'; FONT-SIZE: 12pt"><u></u><u></u></span> </p>
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