Sure. Put all this information into <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a>:<br><br>Your topology, including the SIP provider, any routers/firewalls (including make & model), phone make & model<br>
Your relevant dialplan<br>Console log of the call, with SIP trace turned on<br><br>You can also email me the .pcap file, but I don't want to review it until you've got all the above information posted so that we can get a good picture of what's happening...<br>
<br>Thanks!<br>-MC<br><br><br><div class="gmail_quote">On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt <span dir="ltr"><<a href="mailto:sean@obscuradigital.com">sean@obscuradigital.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div>
<font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;">Hey MC, would you be willing to look over my tcpdump to help determine where the failure is happening?<br><font color="#888888">
<br>
Sean</font><div><div></div><div class="h5"><br>
<br>
<br>
On 6/14/10 12:32 PM, "Michael Collins" <<a href="http://msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>> wrote:<br>
<br>
</div></div></span></font><blockquote><div><div></div><div class="h5"><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;">Hop on the console, turn on siptrace and watch the call flow for clues. You might need to do a tcpdump capturing both signaling and media and then analyze in Wireshark to see what exactly is happening.<br>
-MC<br>
<br>
On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt <<a href="http://sean@obscuradigital.com" target="_blank">sean@obscuradigital.com</a>> wrote:<br>
</span></font><blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;">Hello list,<br>
<br>
I’ve been dealing with a particular issue with in-coming calls. <br>
Leg A calls into the office, then Leg B (endpoint) picks up call but hears nothing on other side. Wait 4 sec call completes and Leg B can hear the other person. <br>
<br>
I have Centos 5.4<br>
Latest git build<br>
Polycom phones<br>
<br>
Calling out is not a problem.<br>
<br>
Not sure how to troubleshoot this issue or maybe there’s a delay setting in the sip profile that waits for the channel to complete.<br>
<br>
Thanks for the help<br>
<font color="#888888">Sean</font> <br>
<br>
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