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. Thanks in advance, Ikka Jakarta, Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160511/28388a8f/attachment.html>
There is no limit as far as asterisk goes. There can be other reasons such as T1
timers or rtptimeout being set. You need to start by enabling sip debug and
seeing who sends the BYE then you need to figure out why they are hanging up.
Regards,
Dovid
-----Original Message-----
From: Ikka Tirtawidjaja <ikka.tirta at gmail.com>
Sender: asterisk-users-bounces at lists.digium.comDate: Wed, 11 May 2016
18:26:48
To: asterisk-users<asterisk-users at lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Subject: [asterisk-users] maximum call time
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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
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 <jcolp at digium.com> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160512/e3a59ca6/attachment.html>