similar to: Phase locked mode

I too had a server room fry and need to replace h/w. So what specific Dell servers did/do you deploy? Where is the link w/Digium/s Dell caveats? I'm using the Digium TDM400 card w/* > Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT) > From: Aaron Daniel <amdtech@shsu.edu> > Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! > To: Asterisk Users Mailing List -

Has anyone successfully gotten two separate servers to handle voicemail on an NFS shared mount? The main thing I'm concerned about at this point is keeping both systems from writing the voicemail file to the same filename... any thoughts? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Does anyone know if asterisk can detect and handle if a phone is forwarded in the dialplan? Aaron

Aaron Any idea how to change it from 24hr to 12hr ? Thanks again! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Hall, Eric M. Sent: Friday, May 05, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question Aaron Yes it

Does anyone have the polycom soundpoint ip's successfully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I know I compiled AEL2 into 1.2 before, considering I just copied my source from one server to another, yet I can't seem to figure out why I'm getting this error. Anyone have any ideas? make[1]: Entering directory `/usr/local/src/asterisk-svn/asterisk/pbx' flex argdesc.l "argdesc.l", line 19: unrecognized %option: reentrant "argdesc.l", line 20: unrecognized

Hey all, Has anyone had any issues with phones having multiple lines that are in different contexts? We've got a couple phones that we're testing intercom functionality for, and I'm noticing that for some strange reason, no matter what line we use, the phones tend to be completely in one context or another, not segregated like I would expect. Our contexts look like this: context

Here's an interesting question: If I transfer a call from Asterisk system to another with IAX, is there any way I can get control back on the original system? Or.. do I lose control, and the dialplan has to continue on the new system? Scenario is we transfer calls to an Asterisk system that handles ACD queues. If the ACD queue times out, we want to send the caller to voicemail on another

In article <Pine.LNX.4.64.0603211635320.7043@ab1-1-246.shsu.edu>, amdtech@shsu.edu says... > You have to set up a dialplan.xml file in your tftpboot directory for the > phone to pull: > > <DIALTEMPLATE> > <TEMPLATE MATCH="9,59....." Timeout="0"/> > <TEMPLATE MATCH="9,29....." Timeout="0"/> >

Of the people in here that have hinting working with the polycom 601's (or any phone for that matter)... do you have it working so that the shared line appearance shows that there's someone on the phone? If so, any hints on how to do it? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Just doing some test installs of asterisk running on branch (noticed first on branch), and noticed if you move to virtual terminal 9 (may be different on everyone else's), the CLI is running. Anyone have any idea how to turn this off? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Anyone have any ideas why DTMF would not work on only one number? Looking through the logs, anytime a button is pressed, this is what shows up: 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Exception on 9, channel 1 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Echo cancellation already on We

I know you can set up monitoring of queued calls, and I'm pretty sure my question's been answered before, but has anyone devised of a way to actually barge into a queue channel so you can do in place monitoring of calls? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate

How does the hinting work on the polycoms? I've got a polycom set up with hinting, I can see when the shared line rings, but I can't tell if someone's on the line. Any suggestions? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Does anyone have the pickup application working? I'm attempting to get it so that a particular extension programmed into a phone can be picked up by another phone with that extension programmed with a speed dial with a 'p' in front... basically, if you dial p100 and extension 100 is ringing, it'll pick up that extension, otherwise it dials the number. The problem I'm having is

Hi, all-- What do you make of this? Here's my call log--looks like there are a lot of calls going in and out of the server that are not real incoming or outgoing calls. Does anybody have any clue what is happening? 2006-11-14 16:41:00 Local/8183... 8183461773 "8183461773" <8183461773> 8183461773 NO ANSWER 1 47. 2006-11-14 16:40:59 IAX2/Voice... 8183461773

Does anyone have any information on how to use DUNDi precaching? Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching, instead of having each DUNDi peer meshed with every other one... http://lists.digium.com/pipermail/dundi/2004-October/000189.html However, it seems that no

Hello all, I first want to thank everyone for all your contributions. I've building an asterisk system for a month or so now and without everyone in the online asterisk community I wouldn't have made it this far yet. Thanks! ...ok, mushiness out of the way.. :) I am looking for a failover and ultimately a load balancing asterisk solution. I've done a good bit of research and I

Anyone know if a way to have voicemail files stored as mp3's? Thanks, Doug.