similar to: Virtual terminal running CLI

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 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

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

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

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

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

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

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

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"/> >

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

Arrgh. I just made a call with Asterisk to extension 2944093. That extension exists in astdb and I have rtcachefriends=yes in sip.conf. Asterisk did a database query... SELECT * FROM ast_sip_users WHERE name = '2944093' Uhm... Why? Doug

Ugh. We thought we'd fixed some problems by using regexten and DUNDi. Guess not. We have a configuration with three Asterisk boxes. Phones register with a single, primary asterisk box under normal conditions. For voicemail deposit, retrieval, we trunk the calls over to our asterisk voicemail server. However, the voicemail server now has no knowledge of the location details of the phones,

Hello Guys, I just got my new Cisco 7940* IP Phone Unfortunately I can't find out how to setup this phone. Am I really that stupid ;-) ? I've read the Cisco Manuel, but everything I tried did not help anything. 1. When I turn on the phone it will display "Configuring VLAN....Configuring IP".. This message will not disappear. 2. I can see that the phone has a local IP. I can

It seems that when you use Realtime static and possibly realtime realtime for sip users, that Asterisk fails to create the regexten context for DUNDi. Someone else had the same problem back in July. Doesn't look like they ever had a resolution. <http://lists.digium.com/pipermail/asterisk-users/2006-July/160105.html>

Hello guys Is there anyway for me to change the sounds that are presented in VoicemailMain? For instance, instead of it saying "mailbox", I would like it to say something like "please enter your mailbox number now". Is there a way for me to do this? I also noticed that when in some of the menus, even if I select one of the announced options it simply repeats the same menu

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