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
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
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
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
--
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
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
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
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
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
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
Hi,
Does anyone know who did the prompts for French and Russian for Asterisk? I
need some custom prompts.
Regards,
Dovid
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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
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
Hi allHow is avg hold time and avg talktime calculated and over long a period of time?Thanks,Israel
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