<br><br><div class="gmail_quote">On Thu, Jun 17, 2010 at 3:38 PM, 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,<br>
Did you by chance get an opportunity to look over my configuration in the pastebin?<br>
<br>
Thanks<br>
Sean<div class="im"><br></div></span></font></div></blockquote><div>Yes. Question - are you deliberately calling four or five different phones when this call comes in? Or do you hve SCA/SLA set up? It looks like a bunch of phones all get rung when this call comes in. Is that what you are trying to do? To make sure there isn't an issue with SCA try sending a call to a phone that is not sharing any of its lines and capture the same information. See if the 4 sec lag is there also. <br>
-MC<br><br></div><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;"><div class="im">
<br>
<br>
On 6/15/10 12:45 PM, "Michael Collins" <<a href="http://msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>> wrote:<br>
<br>
</div></span></font><blockquote><div class="im"><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;">Pastebin, please.<br>
<br>
On Tue, Jun 15, 2010 at 12:19 PM, Sean Holt <<a href="http://sean@obscuradigital.com" target="_blank">sean@obscuradigital.com</a>> wrote:<br>
</span></font></div><blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><div class="im">Should I put the freeswitch log into the pastebin or attach it?<br>
<br>
<br></div>
On 6/15/10 11:11 AM, "Michael Collins" <<a href="http://msc@freeswitch.org" target="_blank">msc@freeswitch.org</a> <<a href="http://msc@freeswitch.org" target="_blank">http://msc@freeswitch.org</a>> > wrote:<br>
<br>
</span></font><blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;">Sure. Put all this information into <a href="http://pastebin.freeswitch.org" target="_blank">pastebin.freeswitch.org</a> <<a href="http://pastebin.freeswitch.org" target="_blank">http://pastebin.freeswitch.org</a>> <<a href="http://pastebin.freeswitch.org" target="_blank">http://pastebin.freeswitch.org</a>> :<div class="im">
<br>
<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><div class="im">
On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt <<a href="http://sean@obscuradigital.com" target="_blank">sean@obscuradigital.com</a> <<a href="http://sean@obscuradigital.com" target="_blank">http://sean@obscuradigital.com</a>> > wrote:<br>
</div></span></font><blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><div class="im">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<br>
</font><br>
<br>
<br></div><div class="im">
On 6/14/10 12:32 PM, "Michael Collins" <<a href="http://msc@freeswitch.org" target="_blank">msc@freeswitch.org</a> <<a href="http://msc@freeswitch.org" target="_blank">http://msc@freeswitch.org</a>> <<a href="http://msc@freeswitch.org" target="_blank">http://msc@freeswitch.org</a>> > wrote:<br>
<br>
</div></span></font><blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><div class="im">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></div><div class="im">
On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt <<a href="http://sean@obscuradigital.com" target="_blank">sean@obscuradigital.com</a> <<a href="http://sean@obscuradigital.com" target="_blank">http://sean@obscuradigital.com</a>> <<a href="http://sean@obscuradigital.com" target="_blank">http://sean@obscuradigital.com</a>> > wrote:<br>
</div></span></font><blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><div class="im">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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