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[rsync@lists.samba.org is a better forum] On 30 Sep 2003, jennifer.a.rawls@ccaix.jsums.edu wrote: > > On 29 Sep 2003, jennifer.a.rawls@ccaix.jsums.edu wrote: > > > Please help... > > > > > > I lock myself out of one of my server... I was testing > > > the rsync -p -e /etc/passwd from my server to another > > > server......

Apologies for asking something that must have been asked many times. I'm running *@home v0.4 and can't get the * time to be local GMT. Tried tzselect etc etc and added ntp server addresses to ntp.conf, * still uses system time of EST so call logs are 5 hours behind. Also, e-mail notifications of vm don't appear to be getting sent, I've set voicemail.conf to include a valid

I'm using a TE405p with all four spans enabled, two configured as pri_cpe and two as pri_net, the asterisk is sitting between our ISDN (UK BT EuroISDN30) and our phone system. We have 200 DDI numbers on the ISDN's and I need to give one of our clients dial-in access to be able to monitor calls we handle on their behalf. I've got ZapBarge working but I need to restrict the client

On Fri, Jan 7, 2011 at 3:56 PM, David Richards <rawdod at gmail.com> wrote: > Its really sad to hear thats happening but even more sad is the fact > that flac is becoming a very common format for music on the interweb > whilst at the same time the development has ceased. I've found some > severe issues with OggFLAC that essentially make it a useless format > for streaming,

I might be able to look into this soon, I actually got my mother a squeezebox boom for xmas, but I have no experience with the device, other than initial set up and hearing it go. My choice of the word "useless" was deliberate to get folks rawled up, and it worked! :D It doesn't make it entirely useless, just unreliable and inconsistent. If they are using the code from the main flac branch unmodified then I'm sure the device is hitting these bugs. There is different bugs in the encoding and decoding side of things, and also in the...

I have a 3xN matrix of parameters obtained from N regressions where the 3 parameters are jointly statistically significant. I would like to reproduce a 3D confidence ellipsoid projecting 2D ellipses onto the XY plane as in Figure 5.2 in this

...RE: RE: 5, 000 concurrent calls system rollout question > (William Boehlke) > 9. Re: ISDN Eicon Diva Server V-BRI (Jens Vagelpohl) > 10. Re: RE: Rewind MusicOnHold? (Dan Journo) > 11. Re: ISDN Eicon Diva Server V-BRI (Armin Schindler) > 12. Agents, queues and zombies (Steve Rawlings) > 13. Fw: Agents, queues and zombies (Steve Rawlings) > 14. Re: ISDN Eicon Diva Server V-BRI (Armin Schindler) > 15. Re: OT O'Reilly Asterisk TFOT (James Ronald) > 16. Re: Blocked Callerid (pdhales@optusnet.com.au) > 17. Re: fax possibilities (pdhales@optusnet.com.au) &...

...awdod at gmail.com> wrote: > I might be able to look into this soon, I actually got my mother a > squeezebox boom for xmas, but I have no experience with the device, > other than initial set up and hearing it go. My choice of the word > "useless" was deliberate to get folks rawled up, and it worked! :D It > doesn't make it entirely useless, just unreliable and inconsistent. If > they are using the code from the main flac branch unmodified then I'm > sure the device is hitting these bugs. There is different bugs in the > encoding and decoding side of thi...

Hi..... I am new to this software, and I want to implement a client (SIP or IAX) with PHP or at least to pass the main functions (connection,call, transfer, hangup, call id etc) to a CRM. Does anyone know if I could achive a project like that with AGI ? Any example using AGI with PHP ? Do I have all the functionality with AGI ? What about call id ? What is depend on ? (As I know * does not

We've been trying unsuccessfully to connect our Meridian Option 81C to a TE110P via PRI. We've followed the directions in asterisk-meridian-a1.pdf (link on http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it doesn't seem to work on our 81C (even though many, many users report it works very well on Option 11's). Has anyone had any success in getting the above

Hi, I'm running *@home 0.5 exactly as it comes off the .iso image and have configured an extension (206) using AMP for a grandstream 102. Have checked sip_additional.conf looks ok, but I can't get the incoming cli on the 102 to read anything other than 't ri', apparently from other posts the phone is trying to display 'asterisk'! I've got a couple of pc's

Apologies for what might seem a lame question but I recently had a set-up where both sides of an agent conversation were combined into a single file using SOX. I've spent hours trying to find this and simply can't, can anyone point me in the right direction, I've tried all sorts of searches at voip-info and here too. I'm running *@home 0.6 if it helps. Steve

Hi all, I'm planning on connecting our Asterisk to our legacy PBX (an Avaya INDeX). I was originally going to sit it between our ISDN connection and the INDeX (tried it, worked ok) but now I intend to hang it off a spare PRI card just so should the * fail we keep our ISDN's at the INDeX, yes I've looked at ISDNguard but little info from the manufacturers. Anyhow, I'm now

Sorry, Forgot to mention, I'm running Asterisk 1.2.3, zaptel and libpri 1.2.2, addons 1.2.1. Steve

Hi all, Have been experimenting with agents and queues instead of placing calls direct to a user's phone extension, but I've run into problems with calls to both the agent and the extension which creates a zombie and double records calls abandoned etc. We're using a unique queue for each agent (only a handful of users) to try and get some agent/queue information to see what the