Displaying 7 results from an estimated 7 matches for "josua".
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joshua
2006 Jun 16
3
Not able to recognize helper class method in controller!
Hi,
Not able to recognize helper class method in controller!
When I try to call some method "get_formatted()" in my controller, it
says local method not recognized.
Please help me out.
Thanks,
josua
--
Posted via http://www.ruby-forum.com/.
2014 Jan 26
2
3g usb dongle - Huawei E1552
...b
Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
So I ran...
[root@]# usb_modeswitch -v 12d1 -p 1446 -W -R -s 60 -c
/etc/usb_modeswitch.d/12d1\:1446
Reading config file: /etc/usb_modeswitch.d/12d1:1446
* usb_modeswitch: handle USB devices with multiple modes
* Version 1.2.3 (C) Josua Dietze 2012
* Based on libusb0 (0.1.12 and above)
! PLEASE REPORT NEW CONFIGURATIONS !
DefaultVendor= 0x12d1
DefaultProduct= 0x1446
TargetVendor= 0x12d1
TargetProduct= not set
TargetClass= not set
TargetProductList="1001,1406,140b,140c,1412,141b,1433,1436,14ac,1506"
DetachStorageOnly=0...
2016 May 12
2
maximum call time
...ut: 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
On Wed, May 11, 2016 at 7:39 PM, Joshua Colp <jcolp at digium.com> wrote:
> Ikka Tirtawidjaja wrote:
>
>> Dear all,
>>
>> is a...
2016 May 11
3
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
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2007 Apr 01
3
Announcement: Asterisk Service Provider Edition v1.0 Beta
...martRTP bridge system, based on our patented VoipRoute core,
makes sure that call latency is minimal. We also enhanced it with a
MediaRescue solution that will capture lost media frames and re-
insert
them in the audio or video stream before it reaches the
destination." says
Josua Polk, the Asterisk RTP developer.
"This system implements an Asterisk VoipRoute layer on top of the
Internet
and uses Dundi(TM) to automatically discover new SmartRTP relays
and their
properties. It practically erases packet loss, jitter and latency
from the list of
issue...
2015 Apr 29
2
PJSIP - sessions-timers support not working on 13.X
Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers
I was getting this :
[Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error:...
2015 Apr 29
0
PJSIP - sessions-timers support not working on 13.X
...how always timers=yes when (timers=no) and (timers=forced) to that endpoint.
I wonder to force asterisk to refresh the session in some cases when is needed .
pjsip is able to refresh the session ?
Cheers
> On Apr 29, 2015, at 1:50 PM, Gosmac <goseeped at gmail.com> wrote:
>
> Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers
>
> I was getting this :
>
> [Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_tex...