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
wrote:
> On Thu, 12 May 2016, Dovid Bender wrote:
>
> Do a simple sip debug and see who sends the bye. You can also simply run
>>
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 and port 5060 -w
/tmp/my-trace.pcap
Or:
sudo ngrep -W byline -d any ^BYE
om>
Subject: Re: [asterisk-users] maximum call time
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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
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
Tirtawidjaja <ikka.ti...@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: [aster
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