asterisk users - May 2016 - maximum call time

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>

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 at gmail.com>
Sender: asterisk-users-bounces at lists.digium.comDate: Thu, 12 May 2016
08:08:49
To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at
lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 <asterisk-users at 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

On Thu, 12 May 2016, Dovid Bender wrote:
> 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
Or:

 	sudo ngrep -W byline -d any ^BYE port 5060

This will display just the BYE messages.

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
             https://www.linkedin.com/in/steve-edwards-4244281