Dear Dovid, thx for the input.
for timer in sip.conf, I used default setting. This is some of the result for "sip show settings" RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) Session Timers: Accept Session Refresher: uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer T1: 500 Timer T1 minimum: 100 Timer B: 32000 Dear Josua, I need to check my server (my settings) first before i complain about it to my provider. Thx to all, Regards, Ikka Jakarta-Indonesia On Wed, May 11, 2016 at 7:39 PM, Joshua Colp <[email protected]> wrote: > Ikka Tirtawidjaja wrote: > >> Dear all, >> >> is asterisk capable to make a call for 24 hour without break ? >> >> My dial string in extension.conf is : >> >> Dial(SIP/[ext_no]@[pbx_name]) >> >> I dont use any dial parameter. >> >> The problemm is, my customer complain that the call was cut after 4 hours. >> > > Providers can also enforce limits to ensure that a call that was not > properly terminated does not result in excess charges. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
