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 <wcse...@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 <l...@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 < <wcse...@selbytech.com> > wcse...@selbytech.com> wrote: > >> On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali < <l...@lopl.net> >> l...@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 >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by <http://www.api-digital.com> >> http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> <http://www.asterisk.org/hello> >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> <http://lists.digium.com/mailman/listinfo/asterisk-users> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Pezhman Lali > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > <http://www.asterisk.org/hello> > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > <http://lists.digium.com/mailman/listinfo/asterisk-users> > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Sherwood McGowan Telecommunications and VOIP Consultant
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