There might be ACK issue. <br>If ACK is not reached to the Phone then it will terminate the call after retransmission timeout.<br>Some time ago I got this issue with SJPhone but not with X-Lite. <br><br clear="all">Santosh Chintalwar<br>
+91 9949695124<br>
<br><br><div class="gmail_quote">On Mon, Jan 18, 2010 at 1:02 AM, Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@asteriskhelpdesk.com">stotaro@asteriskhelpdesk.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br><br><div class="gmail_quote"><div class="im">On Sun, Jan 17, 2010 at 1:05 PM, Olle E. Johansson <span dir="ltr"><<a href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
17 jan 2010 kl. 17.29 skrev Artem Makhutov:<br>
<div><div></div><div><br>
> Hello,<br>
><br>
> I am writing chan_datacard to be able to use Huawei UMTS datacards as a GSM gateway with Asterisk.<br>
><br>
> Some users have complained that their calls are disconnected after exactly 30 seconds.<br>
><br>
> After some investigation it came out that they were using X-Lite as their sofphone and that X-Lite is disconnecting calls after 30 seconds if it does not receive RTCP packages in that time.<br>
><br>
> So the question is how can I get asterisk to send RTCP packets to X-Lite?<br>
><br>
> Must I add something to chan_datacard to generate RTCP packets or should asterisk generate RTCP packets by its own?<br>
> I have no clue about RTCP. Can somebody give me some hints at what I should look at?<br>
<br>
</div></div>Asterisk generates RTCP unless there's an p2p RTP bridge, which I guess won't happen if you place a call from a SIP channel to your own channel. Turn on "rtcp debug" in the CLI and you'll get detailed information about RTCP messaging.<br>
<br>
I have never heard anything about X-lite disconnecting because there's no RTCP reports, that's very odd.<br>
<br>
For a bit more information about RTCP, I suggest that you read my blog about an on-going project to enhance RTCP support in Asterisk:<br>
<a href="http://www.voip-forum.com/asterisk/2010-01/measuring-voice-quality-asterisk/" target="_blank">http://www.voip-forum.com/asterisk/2010-01/measuring-voice-quality-asterisk/</a><br>
<br>
/O<br>
<font color="#888888">--</font></blockquote></div><div><font color="#888888"><br>Not sure if it is related, but 30 seconds disconnect rings a bell if you don't Answer() in some circumstances.<br><br>Thanks,<br>Steve T<br>
</font> </div>
</div>
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