Thanks Michael. I'll go through the resources you mentioned.<div><br></div><div>Thanks,</div><div>Vinuth.</div><div><br><br><div class="gmail_quote">On Tue, Oct 13, 2009 at 2:15 AM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><br><br><div class="gmail_quote"><div class="im">On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur <span dir="ltr"><<a href="mailto:vinuth.madinur@gmail.com" target="_blank">vinuth.madinur@gmail.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">
Hi,<div><br></div><div>Does Freeswitch detect all of these hangup cases mentioned here [<a href="http://wiki.freeswitch.org/wiki/Hangup_causes" target="_blank">http://wiki.freeswitch.org/wiki/Hangup_causes</a>] when using it through a SIP Trunk provider?</div>
<div><br></div><div>If not, should I put in tone_detect application in the dialplan for detecting the SITs? </div><div><br></div><div>Won't freeswitch have to depend on the SIP status sent from SIP trunk to know the hangup status? So, I'm wondering if tone_detect will work at all?</div>
<div><br></div></blockquote></div><div><br>Vinuth,<br><br>As usual, "it depends." Your provider is the key to this whole operation. If the SIP provider sends the information inband then you will definitely need to use tone_detect to look for the SIT tones. However, if the information comes back with the normal SIP messages then you're good to go. I've seen more than a few SIP providers do both, which means that you have to prepare for both cases. <br>
<br>My advice to you is to get pcaps of failed calls and analyze them with Wireshark. If you need help analyzing them then put your pcaps on a web server and post a link so that others can download them. The wiki has some information on grabbing pcaps:<br>
<br><a href="http://wiki.freeswitch.org/wiki/Packet_Capture" target="_blank">http://wiki.freeswitch.org/wiki/Packet_Capture</a><br><br>If you haven't already done so, go to <a href="http://cluecon.com" target="_blank">cluecon.com</a> and download the torrent file that has the ClueCon speaker presentations. The last presentation on Day 3 is Jason Garland and he walks you through using Wireshark for analyzing a SIP call, including both the signaling (SIP) and the media (RTP) parts of the call. BTW, if you have a copy of "VoIP Deployment For Dummies" it has a small section on using Wireshark for call analysis.<br>
<br>-MC<br><br></div></div><br>
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