Displaying 20 results from an estimated 10000 matches similar to: "NFS and voicemail"
2006 Apr 14
1
Re: Asterisk-Users Digest, Vol 21, Issue 81
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 -
2006 Mar 23
7
[OT] Polycom provisioning
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
2006 Nov 03
2
AEL2 in 1.2
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
2006 Apr 03
2
Hinting
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
2006 Apr 11
1
Virtual terminal running CLI
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
2006 Apr 13
1
DTMF Not working for only one number
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
2006 May 23
1
Monitoring queues
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
2006 May 05
1
Spam? Re: Cisco 7970 running SIP question
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
2006 Jun 01
5
Converting Voicemail wav to mp3
Anyone know if a way to have voicemail files stored as mp3's?
Thanks,
Doug.
2006 Mar 24
11
Transferring a call with IAX
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
2006 Mar 27
1
Polycoms and hints
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
2006 May 12
4
DUNDi and Voicemail
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,
2006 Mar 26
1
Re: Cisco 7960 - Have to press a menu button to dial
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"/>
>
2006 Nov 09
5
DUNDi precache
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
2006 Mar 24
8
Asterisk Failover without SER
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
2006 Mar 22
2
Realtime Query
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
2006 Dec 13
1
Pickup application
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
2006 Oct 23
0
Multiple line phones with different contexts
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
2006 Jun 07
2
Unlock / install of Cisco 7940 IP Phone ?
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
2006 Nov 01
2
Realtime, DUNDi and regexten
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>