[asterisk-users] maximum call time

[Dovid Bender]

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