similar to: Dialplan - busy and unavailable without priority jumping

Hi, I'm planning to upgrade an old public/internal development infrastructure and will use CentOS 5.4 x86_64 as basis. One of the critical server is Subversion (as an Apache httpd module). We currently use Subversion v1.4 on the server and v1.6 on our clients. While I was very happy with the v1.4 server over the years, we now sometimes have weird issues which seem related to compatibility

On Wed, Jun 04, 2014 at 02:25:58PM +0530, abhishek jain wrote: > Hi RIch > > > Below are the details of the powerpc ubuntu system ... > > ls -l /boot/ > total 10200 > -rw-r--r-- 1 root root 5224765 Nov 22 2013 uImage > -rw-r--r-- 1 root root 5192861 Nov 22 2013 uImage.orig > > > ls -l /lib/modules/ > total 4 > drwxr-xr-x 3 ubuntu ubuntu 4096 Jun 3

On Wed, Jun 04, 2014 at 03:29:11PM +0530, abhishek jain wrote: > supermin: kernel: SUPERMIN_KERNEL environment variable /boot/uImage > supermin: failed to find a suitable kernel (host_cpu=powerpc64). > > I looked for kernels in /boot and modules in /lib/modules. If you follow the code in src/kernel.ml you'll see that this means supermin was unable to get the kernel version from

Hello shorewall-users, now that I''ve got v1.4 problems solved I''d just like to ask a general question. Are there any real benefits to upgrading if v1.4 does what I want ? I''m not a fan of bleeding-edge in production and I don''t go for "v2 must be better than v1 because it''s newer" Tom, if you have a few minutes, what''s new in 2.0

Hi All, A couple of weeks ago I noticed Askozia PBX, which is a new embedded Asterisk & OS distro at http://askozia.com/pbx. This caught my attention for two reasons; it uses v1.4 of Asterisk, and it uses the m0n0wall development framework to build on FreeBSD with a PHP based GUI. I've used m0n0wall for years, and FreeNAS also, which shares the same OS/GUI framework. I booted the latest

On Wed, Jun 04, 2014 at 01:18:47PM +0530, abhishek jain wrote: > Hi RIch > > Below are the necessary details of the system.. > > > *ls -l /boot/total 10200-rw-r--r-- 1 root root 5224765 Nov 22 2013 > uImage-rw-r--r-- 1 root root 5192861 Nov 22 2013 uImage.origls -l > /lib/modules/total 4drwxr-xr-x 3 ubuntu ubuntu 4096 Jun 3 07:21 > 3.8.13-rt9-QorIQ-SDK-V1.4uname

Hi All, Eagerly waiting for v1.4.x to mature a bit before getting serious about it. Is it ready for production yet? If that's too general, where is it in terms of stability compared to where 1.2.x is now. Anyone running it successfully in production environment and if so what sort of config do you have?

Hello! I can't compile rtmpdump source on FreeBSD: http://sourceforge.net/projects/rtmpdump/ I obtain the following output: # gmake g++ -Wall -c -o bytes.o bytes.cpp In file included from bytes.cpp:25: bytes.h:37:20: endian.h: No such file or directory bytes.h:38:22: byteswap.h: No such file or directory bytes.h:45:2: #error "Undefined byte and float word order!" bytes.cpp:

I recently upgraded my Asterisk system to the latest CVS-HEAD Asterisk CVS HEAD built by bill@localhost on a i686 running Linux on 2005-10-12 13:34:09 UTC Ever since this upgrade, the system is jumping n+101 if it gets a busy on a Dial command, it is now proceeding to the next priority (n+1) Has something changed with this? Is there a way to change it back? Thanks, Bill --------------

Does the stable version of GlusterFS, GlusterFS 1.3-SUSKE RELEASE, have the High Availably (HA) Translator or is it just in v1.4? John This email was independently scanned for viruses by McAfee anti-virus software and none were found

Hi, We have a scenario where we receive calls from 2 different places: 1- Avaya IP Office 2- CIC Interactive Intelligence PBX and the calls are transfer automatically to an Asterisk Box. The problem we are experiencing is that more that half of those calls come with Echo and Jitter. For outbounds calls everything works perfect. I'm on RH9 Stable v1.4 Version, using an quadspad TMD400P digium

Here is a log from a recent call made out on a ZAP channel from a SIP phone inside my network. For some reason, CDR is billing time even though the "busy tone" was detected. It's also logging the call as ANSWERED. Is this normal behavior? Seems a little odd to me. I have this as the first 3 lines of my zapata.conf [channels] busydetect=1 busycount=3 CVS HEAD updated late

I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu --------Dial 1 for support | Dial 2 for special | Dial 3 sales

I'm having problems with ADS membership for samba. I had a "mostly" working version with RHES v2.1, krb5 v1.2, samba v3.0.5. I knew to get to a fully functioning version I would need krb5 v1.3 or later. So finally I had an opertunity to junk RH's crufty krb5 and build from scratch with: RHES v2.1 MIT krb5 v1.4 samba v3.0.13 This works fine on another server. Now to the

I have four database servers in a high-availability, load-balancing configuration. Each machine has a mount to a common data source which is an OCFS2 v1.4 file-system. While working on three of the servers, I restarted the IP network and found after-wards the fourth machine hung. I could not reboot and could not unmount the ocfs2 partitions. I am pretty sure this was all caused by my taking down

I trying to receive a SIP call and have ring a analog phone. I have a TDM11B card with FXS(green) module in line 1. I have Sip server "SER" setup to accept a SIP call, add a 970 extension to uri and set to asterisk SIP server on port 5065. When I send a SIP call from "kphone a soft SIP phone" running to sip://wally.world@cci.net "SER" picks call ok and changes uri

Ok I have spent the last week working on getting my small PBX to work. I will in the end only have 4 SIP extensions being either softphones of IP phones. Currently only 1 SIP config for testing. And at the this point it should be all fairly easy with all inbound and outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via IAX. Inbound does work in it's current basic state.

Thanks for reply. when i try richcomm_usb or blazer_usb or other usb i get the output of : sudo /lib/nut/usbhid-ups -a trust -DD this is the output: Network UPS Tools - Generic HID driver 0.37 (2.6.4) USB communication driver 0.32 Error: UPS [trust] is for driver richcomm_usb, but I'm usbhid-ups! i tried to set the ups.conf to: [trust] driver = usbhid-ups port = auto vendorid = 0925

Chaps, I recently added an incoming VOIP account to my asterisk box. When the PSTN number associated with this account is dialled, the call rings once and then asterisk starts playing music on hold, even though all the extensions continue to ring. Variations of answer() and ringing() don't seem to help. I'm sure I'm missing something spectacularly obvious, but the wiki and googling the

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes