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
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,FreeSWITCH-users@lists.freeswitch.org <http://FreeSWITCH-users@lists.freeswitch.org>
I’ve 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 there’s a delay setting in the sip profile that waits for the channel to complete.
Thanks for the help
Sean
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