Should I put the freeswitch log into the pastebin or attach it? On 6/15/10 11:11 AM, "Michael Collins" <msc@freeswitch.org> wrote:
Sure. Put all this information into pastebin.freeswitch.org <http://pastebin.freeswitch.org> :
Your topology, including the SIP provider, any routers/firewalls (including make & model), phone make & model Your relevant dialplan Console log of the call, with SIP trace turned on
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...
Thanks! -MC
On Tue, Jun 15, 2010 at 10:03 AM, Sean Holt <sean@obscuradigital.com> wrote:
Hey MC, would you be willing to look over my tcpdump to help determine where the failure is happening?
Sean
On 6/14/10 12:32 PM, "Michael Collins" <msc@freeswitch.org <http://msc@freeswitch.org> > wrote:
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. -MC
On Sun, Jun 13, 2010 at 6:14 PM, Sean Holt <sean@obscuradigital.com <http://sean@obscuradigital.com> > wrote:
Hello list,
I¹ve been dealing with a particular issue with in-coming calls. 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.
I have Centos 5.4 Latest git build Polycom phones
Calling out is not a problem.
Not sure how to troubleshoot this issue or maybe there¹s a delay setting in the sip profile that waits for the channel to complete.
Thanks for the help Sean
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