similar to: DTMF Not working for only one number

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 -

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

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

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

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

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

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

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

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"/> >

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

Has anyone successfully gotten an SCCP cisco phone to register to two different asterisk servers at the same time for redundancy? I can't seem to get the phone to recognize the second server in the callmanagergroup... Aaron -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Ok, so multiple people have said that hinting is possible with chan_sccp on the 7940/7960's and such, has anyone got this working? How do you go about getting this to work? I'd use the wiki, but it's link to the mailing list topic on that doesn't work anymore :( -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Has anyone successfully gotten the 7941G working on Asterisk? We're looking at getting some of those instead of the 7940's, but there's really not much info out there about them. -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Aaron Yes it is very annoying! Thanks for the date time settings. That worked GREAT!!! Thanks - Eric -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aaron Daniel Sent: Friday, May 05, 2006 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running

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

Thanks to the great support at Rhino Equipment and Digium, this has finally been solved. I wanted to post the solution back to the list in case anyone else is having a similiar issue. I started by calling Rhino support so I could eliminate channel bank configuration as the issue. We were able to determine the channel bank and signalling were all working as expected. I then began to monitor

Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/ -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I'm currently running a setup that's worked great with MySQL in the past, but we're migrating to ODBC for when we actually integrate the system into our current legacy/voip network. The database I'm trying to use is PostgreSQL, and I've got it working great for Realtime, and if I use cdr_pgsql for the cdr records. However, if I try to use cdr_odbc against the same