search for: josua

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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 -------------- next part -------------- An HTML attachment was scrubbed... URL:
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...