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<TITLE>Re: [Freeswitch-users] 4 second delay</TITLE>
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<FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>I am deliberately calling 5 phones, in a ring group. Previously I did try a single phone, but same issue. I don’t have shared line enables.<BR>
<BR>
Sean<BR>
<BR>
<BR>
On 6/17/10 3:50 PM, "Michael Collins" <<a href="msc@freeswitch.org">msc@freeswitch.org</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'><BR>
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On Thu, Jun 17, 2010 at 3:38 PM, Sean Holt <<a href="sean@obscuradigital.com">sean@obscuradigital.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><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<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>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>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'><BR>
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On 6/15/10 12:45 PM, "Michael Collins" <<a href="msc@freeswitch.org">msc@freeswitch.org</a> <<a href="http://msc@freeswitch.org">http://msc@freeswitch.org</a>> > wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>Pastebin, please.<BR>
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On Tue, Jun 15, 2010 at 12:19 PM, Sean Holt <<a href="sean@obscuradigital.com">sean@obscuradigital.com</a> <<a href="http://sean@obscuradigital.com">http://sean@obscuradigital.com</a>> > wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>Should I put the freeswitch log into the pastebin or attach it?<BR>
<BR>
<BR>
On 6/15/10 11:11 AM, "Michael Collins" <<a href="msc@freeswitch.org">msc@freeswitch.org</a> <<a href="http://msc@freeswitch.org">http://msc@freeswitch.org</a>> <<a href="http://msc@freeswitch.org">http://msc@freeswitch.org</a>> > wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>Sure. Put all this information into pastebin.freeswitch.org <<a href="http://pastebin.freeswitch.org">http://pastebin.freeswitch.org</a>> <<a href="http://pastebin.freeswitch.org">http://pastebin.freeswitch.org</a>> <<a href="http://pastebin.freeswitch.org">http://pastebin.freeswitch.org</a>> :<BR>
<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>
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On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt <<a href="sean@obscuradigital.com">sean@obscuradigital.com</a> <<a href="http://sean@obscuradigital.com">http://sean@obscuradigital.com</a>> <<a href="http://sean@obscuradigital.com">http://sean@obscuradigital.com</a>> > wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><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<BR>
</FONT><BR>
<BR>
<BR>
On 6/14/10 12:32 PM, "Michael Collins" <<a href="msc@freeswitch.org">msc@freeswitch.org</a> <<a href="http://msc@freeswitch.org">http://msc@freeswitch.org</a>> <<a href="http://msc@freeswitch.org">http://msc@freeswitch.org</a>> <<a href="http://msc@freeswitch.org">http://msc@freeswitch.org</a>> > wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><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="sean@obscuradigital.com">sean@obscuradigital.com</a> <<a href="http://sean@obscuradigital.com">http://sean@obscuradigital.com</a>> <<a href="http://sean@obscuradigital.com">http://sean@obscuradigital.com</a>> <<a href="http://sean@obscuradigital.com">http://sean@obscuradigital.com</a>> > wrote:<BR>
</SPAN></FONT></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></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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