Maximum Call Time
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
6 thoughts on - Maximum Call Time
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—
Ikka Tirtawidjaja wrote:
Providers can also enforce limits to ensure that a call that was not properly terminated does not result in excess charges.
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
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 and port 5060 -w /tmp/my-trace.pcap
Regards,
Dovid
—–Original Message—
Or:
sudo ngrep -W byline -d any ^BYE port 5060
This will display just the BYE messages.
Thx for the input. I will try at next time we try to call my pbx for more then 4 hour.