search for: johnathan

...nd 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 and the quality? Thanks Johnathan Falk Network Administrator Clinton Community Schools -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060109/7fb25919/attachment.htm

...ppening after they came up from the meltdown a couple weeks ago. Is caller ID blocking implemented by sending the cid information anyway, but with a bit that says "don't give to end user?" I guess BV would be ignoring this bit. Anyone else experience this with BV and Asterisk? -Johnathan

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...

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

...reconfigured, still has the problem. Outbound calls from the SPA841s through the * server work fine. How do I figure out what the SPAs are unhappy enough about to return 404? Below is a representative SIP DEBUG trace for a call; the OPTION packet sent due to qualify=yes has the same response. -Johnathan Reliably Transmitting (no NAT) to 192.168.1.30:5060: INVITE sip:192.168.1.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4b487795;rport From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as24a55bd8 To: <sip:192.168.1.30> Contact: <sip:asterisk@192.168.1.2> Cal...

...o be problematic. Maybe this is what a "roaming profile" or "trusted domain" is, but I'm not sure. So here's my question: How can I set Samba up to accept logins on one domain with credentials from another, or is this even what I would need to get this working? Thanks, Johnathan -- Johnathan Bell Internet System Administrator, Baker College

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...

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

...bles repeatedly. Going a step further, it seems that if I could parse my original survey file (which is XML) I could get the data I need for most of the labels, and that could be input to the R program, along with the survey data itself. Any suggestions for a good starting point? I'm reading Johnathan Brown's contributed document, "Notes on the use of R for psychology experiments and questionnaires" which is a little helpful, but not quite on target for me. I'm looking to do something in R that's more powerful than a spreadsheet program, and more easily repeatable, but not...

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

...se 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! ===================================================== Johnathan Proffer Viable Technologies, Inc. =====================================================

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