search for: johnathans

Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost

I can call my Broadvoice DID from a outbound caller-id blocked phone, and BV happily delivers the CID to Asterisk (and then on to my IP phone display.) I've tested with the *67 prefix from a PSTN phone to make sure it was supposed to be blocked. The number is always correct, but sometimes the the caller ID name is set to something funky (like a CO or switch center name.) I *think* this

Hey has anyone seen a 1.6 cheatsheet around? Johnathan Snook did a nice 1.5 one but I''ve been working with 1.6 for a while and while I can use prototypejs.org, cheatsheets are handy for jogging memories.. I checked his blog, nothing there for 1.6. Gareth --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups

We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a "you must first dial a 1 or 0" message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six

I don't know what's changed, but four SPA841s and a SPA3000 are no longer answering when they get an inbound call from *. This has been a working configuration for weeks. I *have* been fiddling with the server config; however, the configuration is under version control and I've reverted everything to exactly how it was when the server was working. Doesn't fix it. I reset one of

First, our current setup. I'm setting up a Samba 3.4 environment to replace our old Samba 3.0.x setup and make ready for Windows 7. There are several different campus locations, and each has it's own Samba PDC. Due to some (possibly poor) decisions made early in the development of the system, instead of giving each campus its own domain name, we used the algorithmic rid base parameter in

Hi everyone, I have already started documented and building this privately. However I refuse to believe that there is not a module for setting up a FreeIPA client server on RHEL to authenticate against an existing IPA authentication server. Has anyone actually already done this? If not I will probable post a link to this email with the puppet module I have built. Regards John -- You

I started out happy as a clam with my new Broadvoice account and asterisk machine. About 10 days ago things began to change. Inbound calling has been down for 2 days. Beyond the "We are currently experiencing in-bound call issues with a carrier partner in some areas. We are aware of the issue and our engineers are working to have it resolved as soon as possible" mantra their

hi, To post, and to reply to a post, i have to goto my email. Now, if there is a web interface to these mailing list, things would be easier.

Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond....) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line

Hi folks, Debian 504 64-bit I found following document; http://www.biostat.wisc.edu/~kbroman/Rintro/ Whether it is the right document for installing Emacs+ESS and R so that R can run on Emacs? TIA B.R. Stephen L

Can someone show me how to set up DUNDi, I will be using it to connect 14 asterisk servers internally. I don't want to use it on the external world. If anyone has any examples of connecting 2 or 3 (if their is a difference) machines in a DUNDi co-operation that would be helpful. Johnathan Falk Network Administrator Clinton Community Schools

Greetings Johnathan. Thank you for the reply. I have run top, and iftop, etc... The only process that is listed is as follows: Tasks: 272 total, 2 running, 270 sleeping, 0 stopped, 0 zombie %Cpu(s): 7.1 us, 18.3 sy, 0.0 ni, 73.8 id, 0.7 wa, 0.0 hi, 0.1 si, 0.0 st KiB Mem : 32679644 total, 402520 free, 9889728 used, 22387396 buff/cache KiB Swap: 0 total, 0 free,

On Sun, May 03, 2015 at 08:25:45PM -0400, Tim Dunphy wrote: > Rather than a yum install. If I install the nrpe package from yum I don't > find a check_nrpe script on the system for some reason! That's because the 'check_nrpe' command isn't in the nrpe package. It's in the nagios-plugins-nrpe package. The executable is installed, along side all other nagios check

I have an odd problem. I am running an ubuntu 10.04 domu that I got from stacklet on my centos 5.5/xen4.0.1 dom0. The problem is that even sitting idle the domu has a 1.05 load with init taking up 99% of that. I have searched around but have not found what is causing the issue. I thought I remembered this being mentioned on the list before but could not find it either. My domu.cfg is as

I'm trying to get a semi-automated approach to processing survey data that I'm getting from mod_survey, an Apache-based web survey tool. It looks like one can do this with enough experience in writing R routines. Before I start from scratch, is there a sample file that shows good techniques for producing labeled graphs and tables from survey data with R? I've looked but not found what

maybe this has been covered before but, i can't find it, has anyone had a problem where outside lines can't use number presses like choose extensions but inside lines can, I am using voicetronix hardware with asterisk and when i call from a station port I hear my greeting and can dial an extension and connect, but if I call in I can here my greeting and pushing buttons does nothing, and

Hello, I have a situation where I need to first check if a previous clip was recorded, and if so, append to it.. Otherwise create a new file.. I'm using Monitor. Monitor automatically calls sox after the call ends.. Is there a way to manually control this process, and instruct sox to append to the destination file, rather than overwrite? Thanks in advance!

Hey, I've come across two interesting problems today. First, when recording long calls using Monitor(), it appears the in and out channels become out of sync. It seems like one channel happens faster or has data missing when sox mixes them together. Digging around, I found MixMonitor, which skips the whole soxmix process. I figured that removing that step could only help. Now it seems that

Hello, I'm trying to do call forwarding based on this: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding In the extensions.conf file I have several context defined (local, longdistance, mobile, international and so on). Each user can be associated with different context (so can make only i.e. local calls). How to set calls forwarding only to numbers that are available in