[asterisk-users] 40sec between dial execution and sending SIP request

[Sherwood McGowan]

Good call Warren, might I add that a great idea would be to set debug and
verbose to 5, change the timestamp format on your logs temporarily to show
HH:mm:ss:ms (don't necessarily need milliseconds, but I'm an accuracy geek),
make sure you have a log that is writing ALL output (except maybe DTMF, but
error, warning, info, debug, verbose are all necessary)....

then do a logger reload and a logger rotate, dial your test call, and then
attach the resulting logfile.

On Tue, May 10, 2011 at 2:28 AM, Warren Selby <wcselby at selbytech.com> wrote:

> Show us the cli trace of the delay.
>
> Thanks,
> --Warren Selby, dCAP
>
> On May 10, 2011, at 2:18 AM, Pezhman Lali <lopl at lopl.net> wrote:
>
> thanks,
> this delay is occurred   on asterisk server, between dial execution and
> "CALLED ....."
>
>
> On Mon, May 9, 2011 at 7:12 PM, Warren Selby < <wcselby at selbytech.com>
> wcselby at selbytech.com> wrote:
>
>> On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali < <lopl at lopl.net>
>> lopl at lopl.net> wrote:
>>
>>> Dear
>>> I have a small pbx with asterisk 1.6.2.16.
>>> I have a funny problem, there is exactly 40sec between dial execution and
>>> sending first invite packet on sip.
>>> do you have any idea where the problem is ?
>>>
>>
>> Check the dial timeout on your phone itself.  What model phone do you
>> have?
>>
>> --
>> Thanks,
>> --Warren Selby, dCAP
>> <http://www.selbytech.com>http://www.selbytech.com
>>
>> --
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>
>
>
> --
> Pezhman Lali
>
>
>  --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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> _____________________________________________________________________
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-- 
Sherwood McGowan
Telecommunications and VOIP Consultant
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