Displaying 20 results from an estimated 20000 matches similar to: "Web based voicemail client"
2005 Jan 24
1
Realtime voicemail question
Group
I'm using realtime for voicemail the it works great.. The only problem
I have is I'm not able to use directory or vmail.cgi
Does anyone have a solution for this problem?
Asterisk CVS-HEAD-01/24/05-07:36:37
RedHat 9.0
Any help would be great!!!!
Thanks
2007 May 19
2
(OT) Anyone Ever Use http://shopfort1.com as a Broker
I have no affiliation with them but if their quotes are accurate then
they provide quite a few options as far as TDM connectivity and realtime
pricing.
If you do not want a phone call from a sales person, give them a BTN
that goes to an IVR or something. They call no matter which box you
click as far as "contact me now" "contact me later" "just window
shopping".
2007 May 28
5
Blindside Web Conferencing
Hello,
We are creating a web-based conferencing application using Asterisk as the
voice conferencing server.
This as an open source project. We are trying to determine if there
is interest of the community and perhaps work together to improve the
application.
Using the web application, you can upload your powerpoint presentation,
manage the participants in the conference thru the web interface
2007 Mar 07
1
auto dialer
Not able to get the auto dialer part of asterisk to work with the zap
channel. It works great with the sip channel. Here is the call file and
the CLI output
Call File
Channel: ZAP/G1/6144994925
MaxRetries: 3
RetryTime: 40
WaitTime: 2
Context: amaxx
Extension: 36652
Priority: 1
CLI Output
Connected to Asterisk SVN-branch-1.4-r57207 currently running on
VoIP-PBX (pid
2007 Mar 07
4
OT Vonage V-Phone Adapter (Possible Hack)
It would be cool to get one of these and see if it can be hacked and
loaded with your favorite SIP or IAX softphone. Looking at the pic, it
looks like the dongle is both a soundcard and memory stick. Heck, I
would be glad to have it if I could get the soundcard to work.
Might as well since it is free after rebate.
http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem
2007 Apr 25
3
FYI
Just been getting lots of failed SIP registrations to a system here.
All coming from Taiwan.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net
Euro Tech News Blog http://eurotechnews.blogspot.com
2007 May 24
13
Bottom line on fax reception
So what is the bottom line? Does it work or not. I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.
What's the bottom line with recent updates on 1.2.x? Is it production ready
for fax? By production ready I mean that it just works all the time and
doesn't need any babysitting. Do I have to worry
2004 Jul 06
3
Cisco 7960 and Voice Mail
I search Google to find how to get the message light to flash on my
Cisco 7960 running (Application Load ID POS3-06-3-00) (Boot Load ID
PC03M030) (DSP Load ID PS03AT38)
All I see is about the sip.conf file witch mine has the mailbox=XXXX but
still no light. Also the messages button does not work.
Any ideas?
2007 Jun 03
2
Chan_mobile issue
Hello,
I just did a fresh svn install of 1.4 trunk everything. Everything
compiles and installs just fine.
When I get to asterisk-addons, I cannot select chan_mobile in
menuselect.
Chan_mobile is not even an option in menuselect for asterisk trunk.
I tried the latest patch which failed in many places but did add an
option for chan_mobile in menuselect for asterisk but it still cannot be
2006 Apr 13
2
app_meetme.so
Hi all,
I'm using Asterisk 1.2.5 and , for some reason, when I install it, the
module app_meetme.so didn't install. Is there some way to download
that module, and add it to asterisk without re-install it?
Thanks in advance
Sebastian
2006 Mar 30
3
Please Help Test Quad PRI Using NFAS
Please help me test my setup by dialing 800.564.0215 and listen to the
queue for a bit. I have a quad port T1 with NFAS setup.
I can dial-out but I cannot dial any 800 numbers (Global Crossing says I
need LDS service and that will be a couple weeks) so I cant test it
myself. I need at least 24 callers to feel comfortable enough that it
is working properly.
Thanks,
Steve Totaro
2007 Apr 12
8
test
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<TITLE>Message</TITLE>
<META content="MSHTML 6.00.2900.3059" name=GENERATOR></HEAD>
<BODY>
<DIV> </DIV></BODY></HTML>
2005 Sep 20
1
ODBC Voicemail WEB Retrieval
Ok.
I was sucessful in installing ODBC storage
I'm using MySQL in the backend as it is. but all my drivers are now ODBC.
I am running asterisk-cvs head as of last night 9/19/05
My question is this... the old voicemail.cgi script that allowed checking
voicemail no longer works etc, and never did work for me without a static
voicemail.conf file.
Anyways.. that aside... how does one retrieve
2007 Jun 06
3
1.4 Zaptel/Sangoma Issues on CentOS
Any ideas? Sangoma support is closed for the evening.
I have the latest Sangoma drivers and Asterisk 1.4 everything installed.
When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up"
repeated over and over. The B channels never come up. There are no
errors in any of the logs, zttool, or the wanpipe tools.
Intense pri debug output:
< Unnumbered frame:
< SAPI:
2004 Mar 31
8
Newbie....
I have a question for the group.
To get this running do I need any Digium Cards? I understand I will
need them to connect to the public phone system. I'm looking at just
using IP Phones or IP Softphones just to test this app.
Thanks for any help you could give.
2006 Feb 16
1
ARI 0.06
ARI (Asterisk Recording Interface) has reached another milestone.
The project is starting to become a full featured user portal and
handle all the common errors that people seem to have. This release
supports:
call monitor page ? new features include column sorting and filter
small duration calls
in addition to the ability to listen
to call monitor
2007 May 31
9
click to call
I have been looking around for examples or code on making a click to call
application for web sites... has anybody had any luck on this topic? Is
there any open source code out ther that could do this?
Regards
AK
2007 Mar 14
3
What happend to voip-info?
Anyone has an idea what happend to voip-info? it stopped working about 24
hours ago.
Nir S
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/23cdc0f6/attachment.htm
2007 Mar 19
4
Queue App - Free agent and waiting calls
<asterisk-users@lists.digium.com>Asterisk 1.4
I have strategy= leastrecent and autofill = yes
I have 2 agents, one is answering a call and the other is free and have some
calls waiting in the queue.
Only when the first agent hangup the second agent receive the first call in
the queue.
It happends some times.
This behavior still happend in 1.4.1 version.
Thanks a lot.
-------------- next
2007 Apr 26
1
Asterisk brute force watcher (was FYI)
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of J. Oquendo
> Sent: Thursday, April 26, 2007 6:47 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Asterisk brute force watcher (was FYI)
>
> Steve Totaro wrote:
> > I suspect that