[Freeswitch-users] 4 second delay

Michael Collins msc at freeswitch.org
Thu Jun 17 15:50:02 PDT 2010


On Thu, Jun 17, 2010 at 3:38 PM, Sean Holt <sean at obscuradigital.com> wrote:

>  Hey MC,
> Did you by chance get an opportunity to look over my configuration in the
> pastebin?
>
> Thanks
> Sean
>
> 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.
-MC


>
> On 6/15/10 12:45 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>
> Pastebin, please.
>
> On Tue, Jun 15, 2010 at 12:19 PM, Sean Holt <sean at obscuradigital.com>
> wrote:
>
> Should I put the freeswitch log into the pastebin or attach it?
>
>
> On 6/15/10 11:11 AM, "Michael Collins" <msc at freeswitch.org <
> http://msc@freeswitch.org> > wrote:
>
> Sure. Put all this information into pastebin.freeswitch.org <
> http://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 at obscuradigital.com <
> http://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 at freeswitch.org <
> http://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 at obscuradigital.com <
> http://sean@obscuradigital.com>  <http://sean@obscuradigital.com> > wrote:
>
> Hello list,
>
> 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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