Re: [asterisk-users] maximum call time

2016-05-11 Thread Ikka Tirtawidjaja
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 >>

Re: [asterisk-users] maximum call time

2016-05-11 Thread Steve Edwards
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

Re: [asterisk-users] maximum call time

2016-05-11 Thread 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 and port 5060 -w /tmp/my-trace.pcap Regards, Dovid -Original Message- From: Ikka Tirtawidjaja

Re: [asterisk-users] maximum call time

2016-05-11 Thread Ikka Tirtawidjaja
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

Re: [asterisk-users] Call File - CPU spikes

2016-05-11 Thread Bryant Zimmerman
I am working on a project that we are seeing a 100% CPU spike when we move 50 calls files to the folder. We are running pjsip and asterisk 13..It holds the spike for several minutes Are there any tunable that may help with this? Thanks Bryant --

[asterisk-users] Early Media Dialplan Issue

2016-05-11 Thread Dan Adkins
Hello all, Our company is working with a third party predictive dialer application that uses Asterisk 10.8.0 as its underlying telephony engine.  For several months, we have had issues with the execution of the dialplan due to early media packets being sent from our SIP provider.  My

Re: [asterisk-users] maximum call time

2016-05-11 Thread Joshua Colp
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

Re: [asterisk-users] maximum call time

2016-05-11 Thread Dovid Bender
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

[asterisk-users] maximum call time

2016-05-11 Thread Ikka Tirtawidjaja
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

Re: [asterisk-users] How is Queue avg holdtime and avg talktime calculated

2016-05-11 Thread Israel Gottlieb
Thanks For reports i dont those numbers but i use it for a wallboard and would like to know what timeperiod is taken into account when mesuring the moving average? On Wed, May 11, 2016 at 1:02 PM, Ishfaq Malik wrote: > > > On 11 May 2016 at 10:59, Ishfaq Malik

Re: [asterisk-users] Switching between Music on Hold streams. [13.8.2]

2016-05-11 Thread Dovid Bender
If you ever figure out AAC in Asterisk for MOH let me know. The ones that I have working is MP3 and MMS. On Mon, May 9, 2016 at 1:18 PM, Jonathan H wrote: > Thanks Joshua and everyone, > > Joshua's solution seems a lot simpler and works well. Only one thing > now - The

[asterisk-users] Russian and French sounds

2016-05-11 Thread Dovid Bender
Hi, Does anyone know who did the prompts for French and Russian for Asterisk? I need some custom prompts. Regards, Dovid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

Re: [asterisk-users] How is Queue avg holdtime and avg talktime calculated

2016-05-11 Thread Ishfaq Malik
On 11 May 2016 at 10:59, Ishfaq Malik wrote: > > > On 11 May 2016 at 10:24, Israel Gottlieb wrote: > >> >> Hi all >> >> How is avg hold time and avg talktime calculated and over long a period >> of time? >> >> Thanks, >> Israel >> >> > Hi Israel > > If you

Re: [asterisk-users] How is Queue avg holdtime and avg talktime calculated

2016-05-11 Thread Ishfaq Malik
On 11 May 2016 at 10:24, Israel Gottlieb wrote: > > Hi all > > How is avg hold time and avg talktime calculated and over long a period of > time? > > Thanks, > Israel > > Hi Israel If you are referring to the output of the queue show command then this is the response I

[asterisk-users] How is Queue avg holdtime and avg talktime calculated

2016-05-11 Thread Israel Gottlieb
Hi allHow is avg hold time and avg talktime calculated and over long a period of time?Thanks,Israel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory