VoIP Mailing List Archives

[ikka.tirta at gmail.com]

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

[dovid at telecurve.com]

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@gmail.com>
Sender: asterisk-users-bounces@lists.digium.comDate: Wed, 11 May 2016 18:26:48
To: asterisk-users<asterisk-users@lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] maximum call time

--
_____________________________________________________________________
-- 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

[jcolp at digium.com]

Ikka Tirtawidjaja wrote:

[ikka.tirta at gmail.com]

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@digium.com (jcolp@digium.com)> wrote:

[dovid at telecurve.com]

Ikka,

Do a simple sip debug and see who sends the bye. You can also simply run tcpdump in a screened session and when the call is done analyze in wireshark.
tcpdump -s0 host <IP of carrier> and port 5060 -w /tmp/my-trace.pcap


Regards,

Dovid

-----Original Message-----
From: Ikka Tirtawidjaja <ikka.tirta@gmail.com>
Sender: asterisk-users-bounces@lists.digium.comDate: Thu, 12 May 2016 08:08:49
To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users@lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] maximum call time

--
_____________________________________________________________________
-- 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

[asterisk.org at sedwar...]

On Thu, 12 May 2016, Dovid Bender wrote:

[ikka.tirta at gmail.com]

Thx for the input. I will try at next time we try to call my pbx for more then 4 hour.

On Thu, May 12, 2016 at 8:43 AM, Steve Edwards <asterisk.org@sedwards.com (asterisk.org@sedwards.com)> wrote: